Threads.cpp revision dd9764290b3c1d801fea9505189cae29db919902
15bc087c573c70c84c6a39946457590b42d392a33Andreas Huber/* 25bc087c573c70c84c6a39946457590b42d392a33Andreas Huber** 35bc087c573c70c84c6a39946457590b42d392a33Andreas Huber** Copyright 2012, The Android Open Source Project 45bc087c573c70c84c6a39946457590b42d392a33Andreas Huber** 55bc087c573c70c84c6a39946457590b42d392a33Andreas Huber** Licensed under the Apache License, Version 2.0 (the "License"); 65bc087c573c70c84c6a39946457590b42d392a33Andreas Huber** you may not use this file except in compliance with the License. 75bc087c573c70c84c6a39946457590b42d392a33Andreas Huber** You may obtain a copy of the License at 85bc087c573c70c84c6a39946457590b42d392a33Andreas Huber** 95bc087c573c70c84c6a39946457590b42d392a33Andreas Huber** http://www.apache.org/licenses/LICENSE-2.0 105bc087c573c70c84c6a39946457590b42d392a33Andreas Huber** 115bc087c573c70c84c6a39946457590b42d392a33Andreas Huber** Unless required by applicable law or agreed to in writing, software 125bc087c573c70c84c6a39946457590b42d392a33Andreas Huber** distributed under the License is distributed on an "AS IS" BASIS, 135bc087c573c70c84c6a39946457590b42d392a33Andreas Huber** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 145bc087c573c70c84c6a39946457590b42d392a33Andreas Huber** See the License for the specific language governing permissions and 155bc087c573c70c84c6a39946457590b42d392a33Andreas Huber** limitations under the License. 165bc087c573c70c84c6a39946457590b42d392a33Andreas Huber*/ 175bc087c573c70c84c6a39946457590b42d392a33Andreas Huber 185bc087c573c70c84c6a39946457590b42d392a33Andreas Huber 195bc087c573c70c84c6a39946457590b42d392a33Andreas Huber#define LOG_TAG "AudioFlinger" 205bc087c573c70c84c6a39946457590b42d392a33Andreas Huber//#define LOG_NDEBUG 0 215bc087c573c70c84c6a39946457590b42d392a33Andreas Huber#define ATRACE_TAG ATRACE_TAG_AUDIO 225bc087c573c70c84c6a39946457590b42d392a33Andreas Huber 235bc087c573c70c84c6a39946457590b42d392a33Andreas Huber#include <math.h> 245bc087c573c70c84c6a39946457590b42d392a33Andreas Huber#include <fcntl.h> 255bc087c573c70c84c6a39946457590b42d392a33Andreas Huber#include <sys/stat.h> 265bc087c573c70c84c6a39946457590b42d392a33Andreas Huber#include <cutils/properties.h> 271b86fe063badb5f28c467ade39be0f4008688947Andreas Huber#include <cutils/compiler.h> 285bc087c573c70c84c6a39946457590b42d392a33Andreas Huber#include <utils/Log.h> 295bc087c573c70c84c6a39946457590b42d392a33Andreas Huber#include <utils/Trace.h> 305bc087c573c70c84c6a39946457590b42d392a33Andreas Huber 315bc087c573c70c84c6a39946457590b42d392a33Andreas Huber#include <private/media/AudioTrackShared.h> 325bc087c573c70c84c6a39946457590b42d392a33Andreas Huber#include <hardware/audio.h> 335bc087c573c70c84c6a39946457590b42d392a33Andreas Huber#include <audio_effects/effect_ns.h> 345bc087c573c70c84c6a39946457590b42d392a33Andreas Huber#include <audio_effects/effect_aec.h> 355bc087c573c70c84c6a39946457590b42d392a33Andreas Huber#include <audio_utils/primitives.h> 36ad0d9c9c39a24b7fbd94e935a5855c9025341929Andreas Huber 37b5f25f005bc1d3ae35f45b58c88345e183dc336dAndreas Huber// NBAIO implementations 381b86fe063badb5f28c467ade39be0f4008688947Andreas Huber#include <media/nbaio/AudioStreamOutSink.h> 39ad0d9c9c39a24b7fbd94e935a5855c9025341929Andreas Huber#include <media/nbaio/MonoPipe.h> 4081e68448f3361eaf8618930471fdc3c21bdf5cbcAndreas Huber#include <media/nbaio/MonoPipeReader.h> 41b5f25f005bc1d3ae35f45b58c88345e183dc336dAndreas Huber#include <media/nbaio/Pipe.h> 421b86fe063badb5f28c467ade39be0f4008688947Andreas Huber#include <media/nbaio/PipeReader.h> 43b5f25f005bc1d3ae35f45b58c88345e183dc336dAndreas Huber#include <media/nbaio/SourceAudioBufferProvider.h> 44ad0d9c9c39a24b7fbd94e935a5855c9025341929Andreas Huber 45eac68baf095aeef54865c28b6888924dc6295cbdAndreas Huber#include <powermanager/PowerManager.h> 46dcb89b3b505522efde173c105a851c412f947178Chong Zhang 47dcb89b3b505522efde173c105a851c412f947178Chong Zhang#include <common_time/cc_helper.h> 48ad0d9c9c39a24b7fbd94e935a5855c9025341929Andreas Huber#include <common_time/local_clock.h> 49ad0d9c9c39a24b7fbd94e935a5855c9025341929Andreas Huber 50ad0d9c9c39a24b7fbd94e935a5855c9025341929Andreas Huber#include "AudioFlinger.h" 51ad0d9c9c39a24b7fbd94e935a5855c9025341929Andreas Huber#include "AudioMixer.h" 52ad0d9c9c39a24b7fbd94e935a5855c9025341929Andreas Huber#include "FastMixer.h" 53ad0d9c9c39a24b7fbd94e935a5855c9025341929Andreas Huber#include "ServiceUtilities.h" 54ad0d9c9c39a24b7fbd94e935a5855c9025341929Andreas Huber#include "SchedulingPolicyService.h" 55ad0d9c9c39a24b7fbd94e935a5855c9025341929Andreas Huber 56ad0d9c9c39a24b7fbd94e935a5855c9025341929Andreas Huber#undef ADD_BATTERY_DATA 57ad0d9c9c39a24b7fbd94e935a5855c9025341929Andreas Huber 58ad0d9c9c39a24b7fbd94e935a5855c9025341929Andreas Huber#ifdef ADD_BATTERY_DATA 59ad0d9c9c39a24b7fbd94e935a5855c9025341929Andreas Huber#include <media/IMediaPlayerService.h> 605bc087c573c70c84c6a39946457590b42d392a33Andreas Huber#include <media/IMediaDeathNotifier.h> 615bc087c573c70c84c6a39946457590b42d392a33Andreas Huber#endif 625bc087c573c70c84c6a39946457590b42d392a33Andreas Huber 632048d0cfccce48be26816dec8711a6691ebff71cAndreas Huber// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 642048d0cfccce48be26816dec8711a6691ebff71cAndreas Huber#ifdef DEBUG_CPU_USAGE 6514f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber#include <cpustats/CentralTendencyStatistics.h> 661228d6b175de8b21787cbe0c6c4bb5642f4d555eChong Zhang#include <cpustats/ThreadCpuUsage.h> 671228d6b175de8b21787cbe0c6c4bb5642f4d555eChong Zhang#endif 682048d0cfccce48be26816dec8711a6691ebff71cAndreas Huber 691228d6b175de8b21787cbe0c6c4bb5642f4d555eChong Zhang// ---------------------------------------------------------------------------- 701228d6b175de8b21787cbe0c6c4bb5642f4d555eChong Zhang 7114f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber// Note: the following macro is used for extremely verbose logging message. In 722048d0cfccce48be26816dec8711a6691ebff71cAndreas Huber// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 735bc087c573c70c84c6a39946457590b42d392a33Andreas Huber// 0; but one side effect of this is to turn all LOGV's as well. Some messages 745bc087c573c70c84c6a39946457590b42d392a33Andreas Huber// are so verbose that we want to suppress them even when we have ALOG_ASSERT 759575c96b6e418914e2ffc6741ecc8d71e3968dbeAndreas Huber// turned on. Do not uncomment the #def below unless you really know what you 761228d6b175de8b21787cbe0c6c4bb5642f4d555eChong Zhang// are doing and want to see all of the extremely verbose messages. 771228d6b175de8b21787cbe0c6c4bb5642f4d555eChong Zhang//#define VERY_VERY_VERBOSE_LOGGING 781228d6b175de8b21787cbe0c6c4bb5642f4d555eChong Zhang#ifdef VERY_VERY_VERBOSE_LOGGING 791228d6b175de8b21787cbe0c6c4bb5642f4d555eChong Zhang#define ALOGVV ALOGV 801228d6b175de8b21787cbe0c6c4bb5642f4d555eChong Zhang#else 811228d6b175de8b21787cbe0c6c4bb5642f4d555eChong Zhang#define ALOGVV(a...) do { } while(0) 821228d6b175de8b21787cbe0c6c4bb5642f4d555eChong Zhang#endif 835bc087c573c70c84c6a39946457590b42d392a33Andreas Huber 840df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Hubernamespace android { 850df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber 867314fa17093d514199fedcb55ac41136a1b31cb3Andreas Huber// retry counts for buffer fill timeout 870df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber// 50 * ~20msecs = 1 second 889b80c2bdb205bc143104f54d0743b6eedd67b14eAndreas Huberstatic const int8_t kMaxTrackRetries = 50; 8981e68448f3361eaf8618930471fdc3c21bdf5cbcAndreas Huberstatic const int8_t kMaxTrackStartupRetries = 50; 907314fa17093d514199fedcb55ac41136a1b31cb3Andreas Huber// allow less retry attempts on direct output thread. 915bc087c573c70c84c6a39946457590b42d392a33Andreas Huber// direct outputs can be a scarce resource in audio hardware and should 925bc087c573c70c84c6a39946457590b42d392a33Andreas Huber// be released as quickly as possible. 9314f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huberstatic const int8_t kMaxTrackRetriesDirect = 2; 94ad0d9c9c39a24b7fbd94e935a5855c9025341929Andreas Huber 959575c96b6e418914e2ffc6741ecc8d71e3968dbeAndreas Huber// don't warn about blocked writes or record buffer overflows more often than this 969575c96b6e418914e2ffc6741ecc8d71e3968dbeAndreas Huberstatic const nsecs_t kWarningThrottleNs = seconds(5); 979575c96b6e418914e2ffc6741ecc8d71e3968dbeAndreas Huber 985bc087c573c70c84c6a39946457590b42d392a33Andreas Huber// RecordThread loop sleep time upon application overrun or audio HAL read error 995bc087c573c70c84c6a39946457590b42d392a33Andreas Huberstatic const int kRecordThreadSleepUs = 5000; 10014f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber 1011098d87e56f726184ae9c8fe903ea4743669f35bRobert Shih// maximum time to wait for setParameters to complete 1021098d87e56f726184ae9c8fe903ea4743669f35bRobert Shihstatic const nsecs_t kSetParametersTimeoutNs = seconds(2); 1031098d87e56f726184ae9c8fe903ea4743669f35bRobert Shih 1041098d87e56f726184ae9c8fe903ea4743669f35bRobert Shih// minimum sleep time for the mixer thread loop when tracks are active but in underrun 10514f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huberstatic const uint32_t kMinThreadSleepTimeUs = 5000; 10614f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber// maximum divider applied to the active sleep time in the mixer thread loop 10714f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huberstatic const uint32_t kMaxThreadSleepTimeShift = 2; 10814f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber 10914f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber// minimum normal mix buffer size, expressed in milliseconds rather than frames 1105bc087c573c70c84c6a39946457590b42d392a33Andreas Huberstatic const uint32_t kMinNormalMixBufferSizeMs = 20; 11114f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber// maximum normal mix buffer size 1125bc087c573c70c84c6a39946457590b42d392a33Andreas Huberstatic const uint32_t kMaxNormalMixBufferSizeMs = 24; 1135bc087c573c70c84c6a39946457590b42d392a33Andreas Huber 1145bc087c573c70c84c6a39946457590b42d392a33Andreas Huber// Whether to use fast mixer 11514f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huberstatic const enum { 1165bc087c573c70c84c6a39946457590b42d392a33Andreas Huber FastMixer_Never, // never initialize or use: for debugging only 1175bc087c573c70c84c6a39946457590b42d392a33Andreas Huber FastMixer_Always, // always initialize and use, even if not needed: for debugging only 118eac68baf095aeef54865c28b6888924dc6295cbdAndreas Huber // normal mixer multiplier is 1 119eac68baf095aeef54865c28b6888924dc6295cbdAndreas Huber FastMixer_Static, // initialize if needed, then use all the time if initialized, 1205bc087c573c70c84c6a39946457590b42d392a33Andreas Huber // multiplier is calculated based on min & max normal mixer buffer size 1215bc087c573c70c84c6a39946457590b42d392a33Andreas Huber FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 1225bc087c573c70c84c6a39946457590b42d392a33Andreas Huber // multiplier is calculated based on min & max normal mixer buffer size 1235bc087c573c70c84c6a39946457590b42d392a33Andreas Huber // FIXME for FastMixer_Dynamic: 12414f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber // Supporting this option will require fixing HALs that can't handle large writes. 12514f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber // For example, one HAL implementation returns an error from a large write, 12614f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber // and another HAL implementation corrupts memory, possibly in the sample rate converter. 12714f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber // We could either fix the HAL implementations, or provide a wrapper that breaks 1285bc087c573c70c84c6a39946457590b42d392a33Andreas Huber // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 1295bc087c573c70c84c6a39946457590b42d392a33Andreas Huber} kUseFastMixer = FastMixer_Static; 13043c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber 13143c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber// Priorities for requestPriority 13243c3e6ce02215ca99d506458f596cb1211639f29Andreas Huberstatic const int kPriorityAudioApp = 2; 13343c3e6ce02215ca99d506458f596cb1211639f29Andreas Huberstatic const int kPriorityFastMixer = 3; 134404fced9bfa8fa423ee210a271ca051ffd1bec13Chong Zhang 135404fced9bfa8fa423ee210a271ca051ffd1bec13Chong Zhang// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 136404fced9bfa8fa423ee210a271ca051ffd1bec13Chong Zhang// for the track. The client then sub-divides this into smaller buffers for its use. 137404fced9bfa8fa423ee210a271ca051ffd1bec13Chong Zhang// Currently the client uses double-buffering by default, but doesn't tell us about that. 138404fced9bfa8fa423ee210a271ca051ffd1bec13Chong Zhang// So for now we just assume that client is double-buffered. 139404fced9bfa8fa423ee210a271ca051ffd1bec13Chong Zhang// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 140dcb89b3b505522efde173c105a851c412f947178Chong Zhang// N-buffering, so AudioFlinger could allocate the right amount of memory. 141dcb89b3b505522efde173c105a851c412f947178Chong Zhang// See the client's minBufCount and mNotificationFramesAct calculations for details. 142dcb89b3b505522efde173c105a851c412f947178Chong Zhangstatic const int kFastTrackMultiplier = 2; 143dcb89b3b505522efde173c105a851c412f947178Chong Zhang 144dcb89b3b505522efde173c105a851c412f947178Chong Zhang// ---------------------------------------------------------------------------- 145dcb89b3b505522efde173c105a851c412f947178Chong Zhang 146dcb89b3b505522efde173c105a851c412f947178Chong Zhang#ifdef ADD_BATTERY_DATA 147dcb89b3b505522efde173c105a851c412f947178Chong Zhang// To collect the amplifier usage 148dcb89b3b505522efde173c105a851c412f947178Chong Zhangstatic void addBatteryData(uint32_t params) { 149dcb89b3b505522efde173c105a851c412f947178Chong Zhang sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 150dcb89b3b505522efde173c105a851c412f947178Chong Zhang if (service == NULL) { 151dcb89b3b505522efde173c105a851c412f947178Chong Zhang // it already logged 152dcb89b3b505522efde173c105a851c412f947178Chong Zhang return; 153dcb89b3b505522efde173c105a851c412f947178Chong Zhang } 154dcb89b3b505522efde173c105a851c412f947178Chong Zhang 155dcb89b3b505522efde173c105a851c412f947178Chong Zhang service->addBatteryData(params); 156dcb89b3b505522efde173c105a851c412f947178Chong Zhang} 15784333e0475bc911adc16417f4ca327c975cf6c36Andreas Huber#endif 158dcb89b3b505522efde173c105a851c412f947178Chong Zhang 159dcb89b3b505522efde173c105a851c412f947178Chong Zhang 16043c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber// ---------------------------------------------------------------------------- 16114f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber// CPU Stats 16243c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber// ---------------------------------------------------------------------------- 16343c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber 1640df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huberclass CpuStats { 1650df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huberpublic: 1660df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber CpuStats(); 1670df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber void sample(const String8 &title); 1680df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber#ifdef DEBUG_CPU_USAGE 1690df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huberprivate: 1700df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 1710df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 172dcb89b3b505522efde173c105a851c412f947178Chong Zhang 173dcb89b3b505522efde173c105a851c412f947178Chong Zhang CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 174dcb89b3b505522efde173c105a851c412f947178Chong Zhang 175dcb89b3b505522efde173c105a851c412f947178Chong Zhang int mCpuNum; // thread's current CPU number 176dcb89b3b505522efde173c105a851c412f947178Chong Zhang int mCpukHz; // frequency of thread's current CPU in kHz 177dcb89b3b505522efde173c105a851c412f947178Chong Zhang#endif 178dcb89b3b505522efde173c105a851c412f947178Chong Zhang}; 179dcb89b3b505522efde173c105a851c412f947178Chong Zhang 180dcb89b3b505522efde173c105a851c412f947178Chong ZhangCpuStats::CpuStats() 181dcb89b3b505522efde173c105a851c412f947178Chong Zhang#ifdef DEBUG_CPU_USAGE 182dcb89b3b505522efde173c105a851c412f947178Chong Zhang : mCpuNum(-1), mCpukHz(-1) 183dcb89b3b505522efde173c105a851c412f947178Chong Zhang#endif 184dcb89b3b505522efde173c105a851c412f947178Chong Zhang{ 185dcb89b3b505522efde173c105a851c412f947178Chong Zhang} 186dcb89b3b505522efde173c105a851c412f947178Chong Zhang 187dcb89b3b505522efde173c105a851c412f947178Chong Zhangvoid CpuStats::sample(const String8 &title) { 188dcb89b3b505522efde173c105a851c412f947178Chong Zhang#ifdef DEBUG_CPU_USAGE 189dcb89b3b505522efde173c105a851c412f947178Chong Zhang // get current thread's delta CPU time in wall clock ns 190dcb89b3b505522efde173c105a851c412f947178Chong Zhang double wcNs; 191dcb89b3b505522efde173c105a851c412f947178Chong Zhang bool valid = mCpuUsage.sampleAndEnable(wcNs); 192dcb89b3b505522efde173c105a851c412f947178Chong Zhang 193dcb89b3b505522efde173c105a851c412f947178Chong Zhang // record sample for wall clock statistics 194dcb89b3b505522efde173c105a851c412f947178Chong Zhang if (valid) { 195dcb89b3b505522efde173c105a851c412f947178Chong Zhang mWcStats.sample(wcNs); 196dcb89b3b505522efde173c105a851c412f947178Chong Zhang } 197dcb89b3b505522efde173c105a851c412f947178Chong Zhang 198dcb89b3b505522efde173c105a851c412f947178Chong Zhang // get the current CPU number 199dcb89b3b505522efde173c105a851c412f947178Chong Zhang int cpuNum = sched_getcpu(); 200dcb89b3b505522efde173c105a851c412f947178Chong Zhang 201dcb89b3b505522efde173c105a851c412f947178Chong Zhang // get the current CPU frequency in kHz 202dcb89b3b505522efde173c105a851c412f947178Chong Zhang int cpukHz = mCpuUsage.getCpukHz(cpuNum); 203dcb89b3b505522efde173c105a851c412f947178Chong Zhang 204dcb89b3b505522efde173c105a851c412f947178Chong Zhang // check if either CPU number or frequency changed 2050df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2060df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber mCpuNum = cpuNum; 2070df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber mCpukHz = cpukHz; 2080df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber // ignore sample for purposes of cycles 2090df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber valid = false; 2100df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber } 2110df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber 2120df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber // if no change in CPU number or frequency, then record sample for cycle statistics 2130df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber if (valid && mCpukHz > 0) { 2140df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber double cycles = wcNs * cpukHz * 0.000001; 2150df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber mHzStats.sample(cycles); 2160df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber } 2170df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber 2183e518fd5d679abb83b654e473ea0fa5f7f16235fMarco Nelissen unsigned n = mWcStats.n(); 2193e518fd5d679abb83b654e473ea0fa5f7f16235fMarco Nelissen // mCpuUsage.elapsed() is expensive, so don't call it every loop 2203e518fd5d679abb83b654e473ea0fa5f7f16235fMarco Nelissen if ((n & 127) == 1) { 2213e518fd5d679abb83b654e473ea0fa5f7f16235fMarco Nelissen long long elapsed = mCpuUsage.elapsed(); 2223e518fd5d679abb83b654e473ea0fa5f7f16235fMarco Nelissen if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2233e518fd5d679abb83b654e473ea0fa5f7f16235fMarco Nelissen double perLoop = elapsed / (double) n; 224ced1c2f8f6c422063092f5cc5c675ccdebb2dc10Chong Zhang double perLoop100 = perLoop * 0.01; 2253e518fd5d679abb83b654e473ea0fa5f7f16235fMarco Nelissen double perLoop1k = perLoop * 0.001; 226ced1c2f8f6c422063092f5cc5c675ccdebb2dc10Chong Zhang double mean = mWcStats.mean(); 2273e518fd5d679abb83b654e473ea0fa5f7f16235fMarco Nelissen double stddev = mWcStats.stddev(); 2280df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber double minimum = mWcStats.minimum(); 2290df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber double maximum = mWcStats.maximum(); 2300df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber double meanCycles = mHzStats.mean(); 2310df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber double stddevCycles = mHzStats.stddev(); 2320df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber double minCycles = mHzStats.minimum(); 2330df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber double maxCycles = mHzStats.maximum(); 2340df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber mCpuUsage.resetElapsed(); 2350df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber mWcStats.reset(); 2360df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber mHzStats.reset(); 2370df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber ALOGD("CPU usage for %s over past %.1f secs\n" 2380df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber " (%u mixer loops at %.1f mean ms per loop):\n" 2390df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2400df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2410df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2420df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber title.string(), 2430df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber elapsed * .000000001, n, perLoop * .000001, 2440df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber mean * .001, 2450df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber stddev * .001, 2460df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber minimum * .001, 2470df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber maximum * .001, 2480df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber mean / perLoop100, 2490df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber stddev / perLoop100, 2500df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber minimum / perLoop100, 2510df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber maximum / perLoop100, 2520df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber meanCycles / perLoop1k, 2530df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber stddevCycles / perLoop1k, 2540df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber minCycles / perLoop1k, 25514f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber maxCycles / perLoop1k); 25614f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber 25714f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber } 25814f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber } 25914f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber#endif 26014f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber}; 26114f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber 26214f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber// ---------------------------------------------------------------------------- 26314f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber// ThreadBase 26414f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber// ---------------------------------------------------------------------------- 26514f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber 26614f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas HuberAudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 26714f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 26814f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber : Thread(false /*canCallJava*/), 26914f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber mType(type), 27014f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 27114f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber // mChannelMask 27214f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber mChannelCount(0), 27314f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 27414f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber mParamStatus(NO_ERROR), 27514f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 27614f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 27714f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber // mName will be set by concrete (non-virtual) subclass 27814f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber mDeathRecipient(new PMDeathRecipient(this)) 27914f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber{ 28014f7672b5d450ed26a06fd3bb3ce045ea78b11b2Andreas Huber} 2810df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber 2820df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas HuberAudioFlinger::ThreadBase::~ThreadBase() 2830df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber{ 2840df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber mParamCond.broadcast(); 2850df36ec3303c2c6bf9b42c07945ac8bd234153f3Andreas Huber // do not lock the mutex in destructor 2865bc087c573c70c84c6a39946457590b42d392a33Andreas Huber releaseWakeLock_l(); 2875bc087c573c70c84c6a39946457590b42d392a33Andreas Huber if (mPowerManager != 0) { 288 sp<IBinder> binder = mPowerManager->asBinder(); 289 binder->unlinkToDeath(mDeathRecipient); 290 } 291} 292 293void AudioFlinger::ThreadBase::exit() 294{ 295 ALOGV("ThreadBase::exit"); 296 // do any cleanup required for exit to succeed 297 preExit(); 298 { 299 // This lock prevents the following race in thread (uniprocessor for illustration): 300 // if (!exitPending()) { 301 // // context switch from here to exit() 302 // // exit() calls requestExit(), what exitPending() observes 303 // // exit() calls signal(), which is dropped since no waiters 304 // // context switch back from exit() to here 305 // mWaitWorkCV.wait(...); 306 // // now thread is hung 307 // } 308 AutoMutex lock(mLock); 309 requestExit(); 310 mWaitWorkCV.broadcast(); 311 } 312 // When Thread::requestExitAndWait is made virtual and this method is renamed to 313 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 314 requestExitAndWait(); 315} 316 317status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 318{ 319 status_t status; 320 321 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 322 Mutex::Autolock _l(mLock); 323 324 mNewParameters.add(keyValuePairs); 325 mWaitWorkCV.signal(); 326 // wait condition with timeout in case the thread loop has exited 327 // before the request could be processed 328 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 329 status = mParamStatus; 330 mWaitWorkCV.signal(); 331 } else { 332 status = TIMED_OUT; 333 } 334 return status; 335} 336 337void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 338{ 339 Mutex::Autolock _l(mLock); 340 sendIoConfigEvent_l(event, param); 341} 342 343// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 344void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 345{ 346 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 347 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 348 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 349 param); 350 mWaitWorkCV.signal(); 351} 352 353// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 354void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 355{ 356 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 357 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 358 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 359 mConfigEvents.size(), pid, tid, prio); 360 mWaitWorkCV.signal(); 361} 362 363void AudioFlinger::ThreadBase::processConfigEvents() 364{ 365 mLock.lock(); 366 while (!mConfigEvents.isEmpty()) { 367 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 368 ConfigEvent *event = mConfigEvents[0]; 369 mConfigEvents.removeAt(0); 370 // release mLock before locking AudioFlinger mLock: lock order is always 371 // AudioFlinger then ThreadBase to avoid cross deadlock 372 mLock.unlock(); 373 switch(event->type()) { 374 case CFG_EVENT_PRIO: { 375 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 376 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); 377 if (err != 0) { 378 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 379 "error %d", 380 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 381 } 382 } break; 383 case CFG_EVENT_IO: { 384 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 385 mAudioFlinger->mLock.lock(); 386 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 387 mAudioFlinger->mLock.unlock(); 388 } break; 389 default: 390 ALOGE("processConfigEvents() unknown event type %d", event->type()); 391 break; 392 } 393 delete event; 394 mLock.lock(); 395 } 396 mLock.unlock(); 397} 398 399void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 400{ 401 const size_t SIZE = 256; 402 char buffer[SIZE]; 403 String8 result; 404 405 bool locked = AudioFlinger::dumpTryLock(mLock); 406 if (!locked) { 407 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 408 write(fd, buffer, strlen(buffer)); 409 } 410 411 snprintf(buffer, SIZE, "io handle: %d\n", mId); 412 result.append(buffer); 413 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 414 result.append(buffer); 415 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 416 result.append(buffer); 417 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 418 result.append(buffer); 419 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 420 result.append(buffer); 421 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 422 result.append(buffer); 423 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 424 result.append(buffer); 425 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 426 result.append(buffer); 427 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 428 result.append(buffer); 429 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 430 result.append(buffer); 431 432 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 433 result.append(buffer); 434 result.append(" Index Command"); 435 for (size_t i = 0; i < mNewParameters.size(); ++i) { 436 snprintf(buffer, SIZE, "\n %02d ", i); 437 result.append(buffer); 438 result.append(mNewParameters[i]); 439 } 440 441 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 442 result.append(buffer); 443 for (size_t i = 0; i < mConfigEvents.size(); i++) { 444 mConfigEvents[i]->dump(buffer, SIZE); 445 result.append(buffer); 446 } 447 result.append("\n"); 448 449 write(fd, result.string(), result.size()); 450 451 if (locked) { 452 mLock.unlock(); 453 } 454} 455 456void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 457{ 458 const size_t SIZE = 256; 459 char buffer[SIZE]; 460 String8 result; 461 462 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 463 write(fd, buffer, strlen(buffer)); 464 465 for (size_t i = 0; i < mEffectChains.size(); ++i) { 466 sp<EffectChain> chain = mEffectChains[i]; 467 if (chain != 0) { 468 chain->dump(fd, args); 469 } 470 } 471} 472 473void AudioFlinger::ThreadBase::acquireWakeLock() 474{ 475 Mutex::Autolock _l(mLock); 476 acquireWakeLock_l(); 477} 478 479void AudioFlinger::ThreadBase::acquireWakeLock_l() 480{ 481 if (mPowerManager == 0) { 482 // use checkService() to avoid blocking if power service is not up yet 483 sp<IBinder> binder = 484 defaultServiceManager()->checkService(String16("power")); 485 if (binder == 0) { 486 ALOGW("Thread %s cannot connect to the power manager service", mName); 487 } else { 488 mPowerManager = interface_cast<IPowerManager>(binder); 489 binder->linkToDeath(mDeathRecipient); 490 } 491 } 492 if (mPowerManager != 0) { 493 sp<IBinder> binder = new BBinder(); 494 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 495 binder, 496 String16(mName)); 497 if (status == NO_ERROR) { 498 mWakeLockToken = binder; 499 } 500 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 501 } 502} 503 504void AudioFlinger::ThreadBase::releaseWakeLock() 505{ 506 Mutex::Autolock _l(mLock); 507 releaseWakeLock_l(); 508} 509 510void AudioFlinger::ThreadBase::releaseWakeLock_l() 511{ 512 if (mWakeLockToken != 0) { 513 ALOGV("releaseWakeLock_l() %s", mName); 514 if (mPowerManager != 0) { 515 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 516 } 517 mWakeLockToken.clear(); 518 } 519} 520 521void AudioFlinger::ThreadBase::clearPowerManager() 522{ 523 Mutex::Autolock _l(mLock); 524 releaseWakeLock_l(); 525 mPowerManager.clear(); 526} 527 528void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 529{ 530 sp<ThreadBase> thread = mThread.promote(); 531 if (thread != 0) { 532 thread->clearPowerManager(); 533 } 534 ALOGW("power manager service died !!!"); 535} 536 537void AudioFlinger::ThreadBase::setEffectSuspended( 538 const effect_uuid_t *type, bool suspend, int sessionId) 539{ 540 Mutex::Autolock _l(mLock); 541 setEffectSuspended_l(type, suspend, sessionId); 542} 543 544void AudioFlinger::ThreadBase::setEffectSuspended_l( 545 const effect_uuid_t *type, bool suspend, int sessionId) 546{ 547 sp<EffectChain> chain = getEffectChain_l(sessionId); 548 if (chain != 0) { 549 if (type != NULL) { 550 chain->setEffectSuspended_l(type, suspend); 551 } else { 552 chain->setEffectSuspendedAll_l(suspend); 553 } 554 } 555 556 updateSuspendedSessions_l(type, suspend, sessionId); 557} 558 559void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 560{ 561 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 562 if (index < 0) { 563 return; 564 } 565 566 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 567 mSuspendedSessions.valueAt(index); 568 569 for (size_t i = 0; i < sessionEffects.size(); i++) { 570 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 571 for (int j = 0; j < desc->mRefCount; j++) { 572 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 573 chain->setEffectSuspendedAll_l(true); 574 } else { 575 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 576 desc->mType.timeLow); 577 chain->setEffectSuspended_l(&desc->mType, true); 578 } 579 } 580 } 581} 582 583void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 584 bool suspend, 585 int sessionId) 586{ 587 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 588 589 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 590 591 if (suspend) { 592 if (index >= 0) { 593 sessionEffects = mSuspendedSessions.valueAt(index); 594 } else { 595 mSuspendedSessions.add(sessionId, sessionEffects); 596 } 597 } else { 598 if (index < 0) { 599 return; 600 } 601 sessionEffects = mSuspendedSessions.valueAt(index); 602 } 603 604 605 int key = EffectChain::kKeyForSuspendAll; 606 if (type != NULL) { 607 key = type->timeLow; 608 } 609 index = sessionEffects.indexOfKey(key); 610 611 sp<SuspendedSessionDesc> desc; 612 if (suspend) { 613 if (index >= 0) { 614 desc = sessionEffects.valueAt(index); 615 } else { 616 desc = new SuspendedSessionDesc(); 617 if (type != NULL) { 618 desc->mType = *type; 619 } 620 sessionEffects.add(key, desc); 621 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 622 } 623 desc->mRefCount++; 624 } else { 625 if (index < 0) { 626 return; 627 } 628 desc = sessionEffects.valueAt(index); 629 if (--desc->mRefCount == 0) { 630 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 631 sessionEffects.removeItemsAt(index); 632 if (sessionEffects.isEmpty()) { 633 ALOGV("updateSuspendedSessions_l() restore removing session %d", 634 sessionId); 635 mSuspendedSessions.removeItem(sessionId); 636 } 637 } 638 } 639 if (!sessionEffects.isEmpty()) { 640 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 641 } 642} 643 644void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 645 bool enabled, 646 int sessionId) 647{ 648 Mutex::Autolock _l(mLock); 649 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 650} 651 652void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 653 bool enabled, 654 int sessionId) 655{ 656 if (mType != RECORD) { 657 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 658 // another session. This gives the priority to well behaved effect control panels 659 // and applications not using global effects. 660 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 661 // global effects 662 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 663 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 664 } 665 } 666 667 sp<EffectChain> chain = getEffectChain_l(sessionId); 668 if (chain != 0) { 669 chain->checkSuspendOnEffectEnabled(effect, enabled); 670 } 671} 672 673// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 674sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 675 const sp<AudioFlinger::Client>& client, 676 const sp<IEffectClient>& effectClient, 677 int32_t priority, 678 int sessionId, 679 effect_descriptor_t *desc, 680 int *enabled, 681 status_t *status 682 ) 683{ 684 sp<EffectModule> effect; 685 sp<EffectHandle> handle; 686 status_t lStatus; 687 sp<EffectChain> chain; 688 bool chainCreated = false; 689 bool effectCreated = false; 690 bool effectRegistered = false; 691 692 lStatus = initCheck(); 693 if (lStatus != NO_ERROR) { 694 ALOGW("createEffect_l() Audio driver not initialized."); 695 goto Exit; 696 } 697 698 // Do not allow effects with session ID 0 on direct output or duplicating threads 699 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 700 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 701 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 702 desc->name, sessionId); 703 lStatus = BAD_VALUE; 704 goto Exit; 705 } 706 // Only Pre processor effects are allowed on input threads and only on input threads 707 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 708 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 709 desc->name, desc->flags, mType); 710 lStatus = BAD_VALUE; 711 goto Exit; 712 } 713 714 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 715 716 { // scope for mLock 717 Mutex::Autolock _l(mLock); 718 719 // check for existing effect chain with the requested audio session 720 chain = getEffectChain_l(sessionId); 721 if (chain == 0) { 722 // create a new chain for this session 723 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 724 chain = new EffectChain(this, sessionId); 725 addEffectChain_l(chain); 726 chain->setStrategy(getStrategyForSession_l(sessionId)); 727 chainCreated = true; 728 } else { 729 effect = chain->getEffectFromDesc_l(desc); 730 } 731 732 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 733 734 if (effect == 0) { 735 int id = mAudioFlinger->nextUniqueId(); 736 // Check CPU and memory usage 737 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 738 if (lStatus != NO_ERROR) { 739 goto Exit; 740 } 741 effectRegistered = true; 742 // create a new effect module if none present in the chain 743 effect = new EffectModule(this, chain, desc, id, sessionId); 744 lStatus = effect->status(); 745 if (lStatus != NO_ERROR) { 746 goto Exit; 747 } 748 lStatus = chain->addEffect_l(effect); 749 if (lStatus != NO_ERROR) { 750 goto Exit; 751 } 752 effectCreated = true; 753 754 effect->setDevice(mOutDevice); 755 effect->setDevice(mInDevice); 756 effect->setMode(mAudioFlinger->getMode()); 757 effect->setAudioSource(mAudioSource); 758 } 759 // create effect handle and connect it to effect module 760 handle = new EffectHandle(effect, client, effectClient, priority); 761 lStatus = effect->addHandle(handle.get()); 762 if (enabled != NULL) { 763 *enabled = (int)effect->isEnabled(); 764 } 765 } 766 767Exit: 768 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 769 Mutex::Autolock _l(mLock); 770 if (effectCreated) { 771 chain->removeEffect_l(effect); 772 } 773 if (effectRegistered) { 774 AudioSystem::unregisterEffect(effect->id()); 775 } 776 if (chainCreated) { 777 removeEffectChain_l(chain); 778 } 779 handle.clear(); 780 } 781 782 if (status != NULL) { 783 *status = lStatus; 784 } 785 return handle; 786} 787 788sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 789{ 790 Mutex::Autolock _l(mLock); 791 return getEffect_l(sessionId, effectId); 792} 793 794sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 795{ 796 sp<EffectChain> chain = getEffectChain_l(sessionId); 797 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 798} 799 800// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 801// PlaybackThread::mLock held 802status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 803{ 804 // check for existing effect chain with the requested audio session 805 int sessionId = effect->sessionId(); 806 sp<EffectChain> chain = getEffectChain_l(sessionId); 807 bool chainCreated = false; 808 809 if (chain == 0) { 810 // create a new chain for this session 811 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 812 chain = new EffectChain(this, sessionId); 813 addEffectChain_l(chain); 814 chain->setStrategy(getStrategyForSession_l(sessionId)); 815 chainCreated = true; 816 } 817 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 818 819 if (chain->getEffectFromId_l(effect->id()) != 0) { 820 ALOGW("addEffect_l() %p effect %s already present in chain %p", 821 this, effect->desc().name, chain.get()); 822 return BAD_VALUE; 823 } 824 825 status_t status = chain->addEffect_l(effect); 826 if (status != NO_ERROR) { 827 if (chainCreated) { 828 removeEffectChain_l(chain); 829 } 830 return status; 831 } 832 833 effect->setDevice(mOutDevice); 834 effect->setDevice(mInDevice); 835 effect->setMode(mAudioFlinger->getMode()); 836 effect->setAudioSource(mAudioSource); 837 return NO_ERROR; 838} 839 840void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 841 842 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 843 effect_descriptor_t desc = effect->desc(); 844 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 845 detachAuxEffect_l(effect->id()); 846 } 847 848 sp<EffectChain> chain = effect->chain().promote(); 849 if (chain != 0) { 850 // remove effect chain if removing last effect 851 if (chain->removeEffect_l(effect) == 0) { 852 removeEffectChain_l(chain); 853 } 854 } else { 855 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 856 } 857} 858 859void AudioFlinger::ThreadBase::lockEffectChains_l( 860 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 861{ 862 effectChains = mEffectChains; 863 for (size_t i = 0; i < mEffectChains.size(); i++) { 864 mEffectChains[i]->lock(); 865 } 866} 867 868void AudioFlinger::ThreadBase::unlockEffectChains( 869 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 870{ 871 for (size_t i = 0; i < effectChains.size(); i++) { 872 effectChains[i]->unlock(); 873 } 874} 875 876sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 877{ 878 Mutex::Autolock _l(mLock); 879 return getEffectChain_l(sessionId); 880} 881 882sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 883{ 884 size_t size = mEffectChains.size(); 885 for (size_t i = 0; i < size; i++) { 886 if (mEffectChains[i]->sessionId() == sessionId) { 887 return mEffectChains[i]; 888 } 889 } 890 return 0; 891} 892 893void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 894{ 895 Mutex::Autolock _l(mLock); 896 size_t size = mEffectChains.size(); 897 for (size_t i = 0; i < size; i++) { 898 mEffectChains[i]->setMode_l(mode); 899 } 900} 901 902void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 903 EffectHandle *handle, 904 bool unpinIfLast) { 905 906 Mutex::Autolock _l(mLock); 907 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 908 // delete the effect module if removing last handle on it 909 if (effect->removeHandle(handle) == 0) { 910 if (!effect->isPinned() || unpinIfLast) { 911 removeEffect_l(effect); 912 AudioSystem::unregisterEffect(effect->id()); 913 } 914 } 915} 916 917// ---------------------------------------------------------------------------- 918// Playback 919// ---------------------------------------------------------------------------- 920 921AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 922 AudioStreamOut* output, 923 audio_io_handle_t id, 924 audio_devices_t device, 925 type_t type) 926 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 927 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 928 // mStreamTypes[] initialized in constructor body 929 mOutput(output), 930 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 931 mMixerStatus(MIXER_IDLE), 932 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 933 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 934 mScreenState(AudioFlinger::mScreenState), 935 // index 0 is reserved for normal mixer's submix 936 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 937{ 938 snprintf(mName, kNameLength, "AudioOut_%X", id); 939 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 940 941 // Assumes constructor is called by AudioFlinger with it's mLock held, but 942 // it would be safer to explicitly pass initial masterVolume/masterMute as 943 // parameter. 944 // 945 // If the HAL we are using has support for master volume or master mute, 946 // then do not attenuate or mute during mixing (just leave the volume at 1.0 947 // and the mute set to false). 948 mMasterVolume = audioFlinger->masterVolume_l(); 949 mMasterMute = audioFlinger->masterMute_l(); 950 if (mOutput && mOutput->audioHwDev) { 951 if (mOutput->audioHwDev->canSetMasterVolume()) { 952 mMasterVolume = 1.0; 953 } 954 955 if (mOutput->audioHwDev->canSetMasterMute()) { 956 mMasterMute = false; 957 } 958 } 959 960 readOutputParameters(); 961 962 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 963 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 964 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 965 stream = (audio_stream_type_t) (stream + 1)) { 966 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 967 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 968 } 969 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 970 // because mAudioFlinger doesn't have one to copy from 971} 972 973AudioFlinger::PlaybackThread::~PlaybackThread() 974{ 975 mAudioFlinger->unregisterWriter(mNBLogWriter); 976 delete [] mMixBuffer; 977} 978 979void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 980{ 981 dumpInternals(fd, args); 982 dumpTracks(fd, args); 983 dumpEffectChains(fd, args); 984} 985 986void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 987{ 988 const size_t SIZE = 256; 989 char buffer[SIZE]; 990 String8 result; 991 992 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 993 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 994 const stream_type_t *st = &mStreamTypes[i]; 995 if (i > 0) { 996 result.appendFormat(", "); 997 } 998 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 999 if (st->mute) { 1000 result.append("M"); 1001 } 1002 } 1003 result.append("\n"); 1004 write(fd, result.string(), result.length()); 1005 result.clear(); 1006 1007 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1008 result.append(buffer); 1009 Track::appendDumpHeader(result); 1010 for (size_t i = 0; i < mTracks.size(); ++i) { 1011 sp<Track> track = mTracks[i]; 1012 if (track != 0) { 1013 track->dump(buffer, SIZE); 1014 result.append(buffer); 1015 } 1016 } 1017 1018 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1019 result.append(buffer); 1020 Track::appendDumpHeader(result); 1021 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1022 sp<Track> track = mActiveTracks[i].promote(); 1023 if (track != 0) { 1024 track->dump(buffer, SIZE); 1025 result.append(buffer); 1026 } 1027 } 1028 write(fd, result.string(), result.size()); 1029 1030 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1031 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1032 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1033 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1034} 1035 1036void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1037{ 1038 const size_t SIZE = 256; 1039 char buffer[SIZE]; 1040 String8 result; 1041 1042 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1043 result.append(buffer); 1044 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1045 ns2ms(systemTime() - mLastWriteTime)); 1046 result.append(buffer); 1047 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1048 result.append(buffer); 1049 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1050 result.append(buffer); 1051 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1052 result.append(buffer); 1053 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1054 result.append(buffer); 1055 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1056 result.append(buffer); 1057 write(fd, result.string(), result.size()); 1058 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1059 1060 dumpBase(fd, args); 1061} 1062 1063// Thread virtuals 1064status_t AudioFlinger::PlaybackThread::readyToRun() 1065{ 1066 status_t status = initCheck(); 1067 if (status == NO_ERROR) { 1068 ALOGI("AudioFlinger's thread %p ready to run", this); 1069 } else { 1070 ALOGE("No working audio driver found."); 1071 } 1072 return status; 1073} 1074 1075void AudioFlinger::PlaybackThread::onFirstRef() 1076{ 1077 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1078} 1079 1080// ThreadBase virtuals 1081void AudioFlinger::PlaybackThread::preExit() 1082{ 1083 ALOGV(" preExit()"); 1084 // FIXME this is using hard-coded strings but in the future, this functionality will be 1085 // converted to use audio HAL extensions required to support tunneling 1086 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1087} 1088 1089// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1090sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1091 const sp<AudioFlinger::Client>& client, 1092 audio_stream_type_t streamType, 1093 uint32_t sampleRate, 1094 audio_format_t format, 1095 audio_channel_mask_t channelMask, 1096 size_t frameCount, 1097 const sp<IMemory>& sharedBuffer, 1098 int sessionId, 1099 IAudioFlinger::track_flags_t *flags, 1100 pid_t tid, 1101 status_t *status) 1102{ 1103 sp<Track> track; 1104 status_t lStatus; 1105 1106 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1107 1108 // client expresses a preference for FAST, but we get the final say 1109 if (*flags & IAudioFlinger::TRACK_FAST) { 1110 if ( 1111 // not timed 1112 (!isTimed) && 1113 // either of these use cases: 1114 ( 1115 // use case 1: shared buffer with any frame count 1116 ( 1117 (sharedBuffer != 0) 1118 ) || 1119 // use case 2: callback handler and frame count is default or at least as large as HAL 1120 ( 1121 (tid != -1) && 1122 ((frameCount == 0) || 1123 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1124 ) 1125 ) && 1126 // PCM data 1127 audio_is_linear_pcm(format) && 1128 // mono or stereo 1129 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1130 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1131#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1132 // hardware sample rate 1133 (sampleRate == mSampleRate) && 1134#endif 1135 // normal mixer has an associated fast mixer 1136 hasFastMixer() && 1137 // there are sufficient fast track slots available 1138 (mFastTrackAvailMask != 0) 1139 // FIXME test that MixerThread for this fast track has a capable output HAL 1140 // FIXME add a permission test also? 1141 ) { 1142 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1143 if (frameCount == 0) { 1144 frameCount = mFrameCount * kFastTrackMultiplier; 1145 } 1146 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1147 frameCount, mFrameCount); 1148 } else { 1149 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1150 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1151 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1152 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1153 audio_is_linear_pcm(format), 1154 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1155 *flags &= ~IAudioFlinger::TRACK_FAST; 1156 // For compatibility with AudioTrack calculation, buffer depth is forced 1157 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1158 // This is probably too conservative, but legacy application code may depend on it. 1159 // If you change this calculation, also review the start threshold which is related. 1160 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1161 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1162 if (minBufCount < 2) { 1163 minBufCount = 2; 1164 } 1165 size_t minFrameCount = mNormalFrameCount * minBufCount; 1166 if (frameCount < minFrameCount) { 1167 frameCount = minFrameCount; 1168 } 1169 } 1170 } 1171 1172 if (mType == DIRECT) { 1173 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1174 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1175 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1176 "for output %p with format %d", 1177 sampleRate, format, channelMask, mOutput, mFormat); 1178 lStatus = BAD_VALUE; 1179 goto Exit; 1180 } 1181 } 1182 } else { 1183 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1184 if (sampleRate > mSampleRate*2) { 1185 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1186 lStatus = BAD_VALUE; 1187 goto Exit; 1188 } 1189 } 1190 1191 lStatus = initCheck(); 1192 if (lStatus != NO_ERROR) { 1193 ALOGE("Audio driver not initialized."); 1194 goto Exit; 1195 } 1196 1197 { // scope for mLock 1198 Mutex::Autolock _l(mLock); 1199 mNBLogWriter->logf("createTrack_l isFast=%d caller=%d", 1200 (*flags & IAudioFlinger::TRACK_FAST) != 0, IPCThreadState::self()->getCallingPid()); 1201 1202 // all tracks in same audio session must share the same routing strategy otherwise 1203 // conflicts will happen when tracks are moved from one output to another by audio policy 1204 // manager 1205 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1206 for (size_t i = 0; i < mTracks.size(); ++i) { 1207 sp<Track> t = mTracks[i]; 1208 if (t != 0 && !t->isOutputTrack()) { 1209 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1210 if (sessionId == t->sessionId() && strategy != actual) { 1211 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1212 strategy, actual); 1213 lStatus = BAD_VALUE; 1214 goto Exit; 1215 } 1216 } 1217 } 1218 1219 if (!isTimed) { 1220 track = new Track(this, client, streamType, sampleRate, format, 1221 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1222 } else { 1223 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1224 channelMask, frameCount, sharedBuffer, sessionId); 1225 } 1226 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1227 lStatus = NO_MEMORY; 1228 goto Exit; 1229 } 1230 mTracks.add(track); 1231 1232 sp<EffectChain> chain = getEffectChain_l(sessionId); 1233 if (chain != 0) { 1234 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1235 track->setMainBuffer(chain->inBuffer()); 1236 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1237 chain->incTrackCnt(); 1238 } 1239 1240 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1241 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1242 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1243 // so ask activity manager to do this on our behalf 1244 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1245 } 1246 } 1247 1248 lStatus = NO_ERROR; 1249 1250Exit: 1251 if (status) { 1252 *status = lStatus; 1253 } 1254 return track; 1255} 1256 1257uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1258{ 1259 return latency; 1260} 1261 1262uint32_t AudioFlinger::PlaybackThread::latency() const 1263{ 1264 Mutex::Autolock _l(mLock); 1265 return latency_l(); 1266} 1267uint32_t AudioFlinger::PlaybackThread::latency_l() const 1268{ 1269 if (initCheck() == NO_ERROR) { 1270 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1271 } else { 1272 return 0; 1273 } 1274} 1275 1276void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1277{ 1278 Mutex::Autolock _l(mLock); 1279 // Don't apply master volume in SW if our HAL can do it for us. 1280 if (mOutput && mOutput->audioHwDev && 1281 mOutput->audioHwDev->canSetMasterVolume()) { 1282 mMasterVolume = 1.0; 1283 } else { 1284 mMasterVolume = value; 1285 } 1286} 1287 1288void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1289{ 1290 Mutex::Autolock _l(mLock); 1291 // Don't apply master mute in SW if our HAL can do it for us. 1292 if (mOutput && mOutput->audioHwDev && 1293 mOutput->audioHwDev->canSetMasterMute()) { 1294 mMasterMute = false; 1295 } else { 1296 mMasterMute = muted; 1297 } 1298} 1299 1300void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1301{ 1302 Mutex::Autolock _l(mLock); 1303 mStreamTypes[stream].volume = value; 1304} 1305 1306void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1307{ 1308 Mutex::Autolock _l(mLock); 1309 mStreamTypes[stream].mute = muted; 1310} 1311 1312float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1313{ 1314 Mutex::Autolock _l(mLock); 1315 return mStreamTypes[stream].volume; 1316} 1317 1318// addTrack_l() must be called with ThreadBase::mLock held 1319status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1320{ 1321 mNBLogWriter->logf("addTrack_l mName=%d mFastIndex=%d caller=%d", track->mName, 1322 track->mFastIndex, IPCThreadState::self()->getCallingPid()); 1323 status_t status = ALREADY_EXISTS; 1324 1325 // set retry count for buffer fill 1326 track->mRetryCount = kMaxTrackStartupRetries; 1327 if (mActiveTracks.indexOf(track) < 0) { 1328 // the track is newly added, make sure it fills up all its 1329 // buffers before playing. This is to ensure the client will 1330 // effectively get the latency it requested. 1331 track->mFillingUpStatus = Track::FS_FILLING; 1332 track->mResetDone = false; 1333 track->mPresentationCompleteFrames = 0; 1334 mActiveTracks.add(track); 1335 if (track->mainBuffer() != mMixBuffer) { 1336 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1337 if (chain != 0) { 1338 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1339 track->sessionId()); 1340 chain->incActiveTrackCnt(); 1341 } 1342 } 1343 1344 status = NO_ERROR; 1345 } 1346 1347 ALOGV("mWaitWorkCV.broadcast"); 1348 mWaitWorkCV.broadcast(); 1349 1350 return status; 1351} 1352 1353// destroyTrack_l() must be called with ThreadBase::mLock held 1354void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1355{ 1356 mNBLogWriter->logTimestamp(); 1357 mNBLogWriter->logf("destroyTrack_l mName=%d mFastIndex=%d mClientPid=%d", track->mName, 1358 track->mFastIndex, track->mClient != 0 ? track->mClient->pid() : -1); 1359 track->mState = TrackBase::TERMINATED; 1360 // active tracks are removed by threadLoop() 1361 if (mActiveTracks.indexOf(track) < 0) { 1362 removeTrack_l(track); 1363 } 1364} 1365 1366void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1367{ 1368 mNBLogWriter->logTimestamp(); 1369 mNBLogWriter->logf("removeTrack_l mName=%d mFastIndex=%d clientPid=%d", track->mName, 1370 track->mFastIndex, track->mClient != 0 ? track->mClient->pid() : -1); 1371 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1372 mTracks.remove(track); 1373 deleteTrackName_l(track->name()); 1374 // redundant as track is about to be destroyed, for dumpsys only 1375 track->mName = -1; 1376 if (track->isFastTrack()) { 1377 int index = track->mFastIndex; 1378 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1379 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1380 mFastTrackAvailMask |= 1 << index; 1381 // redundant as track is about to be destroyed, for dumpsys only 1382 track->mFastIndex = -1; 1383 } 1384 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1385 if (chain != 0) { 1386 chain->decTrackCnt(); 1387 } 1388} 1389 1390String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1391{ 1392 String8 out_s8 = String8(""); 1393 char *s; 1394 1395 Mutex::Autolock _l(mLock); 1396 if (initCheck() != NO_ERROR) { 1397 return out_s8; 1398 } 1399 1400 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1401 out_s8 = String8(s); 1402 free(s); 1403 return out_s8; 1404} 1405 1406// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1407void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1408 AudioSystem::OutputDescriptor desc; 1409 void *param2 = NULL; 1410 1411 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1412 param); 1413 1414 switch (event) { 1415 case AudioSystem::OUTPUT_OPENED: 1416 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1417 desc.channels = mChannelMask; 1418 desc.samplingRate = mSampleRate; 1419 desc.format = mFormat; 1420 desc.frameCount = mNormalFrameCount; // FIXME see 1421 // AudioFlinger::frameCount(audio_io_handle_t) 1422 desc.latency = latency(); 1423 param2 = &desc; 1424 break; 1425 1426 case AudioSystem::STREAM_CONFIG_CHANGED: 1427 param2 = ¶m; 1428 case AudioSystem::OUTPUT_CLOSED: 1429 default: 1430 break; 1431 } 1432 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1433} 1434 1435void AudioFlinger::PlaybackThread::readOutputParameters() 1436{ 1437 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1438 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1439 mChannelCount = (uint16_t)popcount(mChannelMask); 1440 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1441 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1442 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1443 if (mFrameCount & 15) { 1444 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1445 mFrameCount); 1446 } 1447 1448 // Calculate size of normal mix buffer relative to the HAL output buffer size 1449 double multiplier = 1.0; 1450 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1451 kUseFastMixer == FastMixer_Dynamic)) { 1452 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1453 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1454 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1455 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1456 maxNormalFrameCount = maxNormalFrameCount & ~15; 1457 if (maxNormalFrameCount < minNormalFrameCount) { 1458 maxNormalFrameCount = minNormalFrameCount; 1459 } 1460 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1461 if (multiplier <= 1.0) { 1462 multiplier = 1.0; 1463 } else if (multiplier <= 2.0) { 1464 if (2 * mFrameCount <= maxNormalFrameCount) { 1465 multiplier = 2.0; 1466 } else { 1467 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1468 } 1469 } else { 1470 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1471 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1472 // track, but we sometimes have to do this to satisfy the maximum frame count 1473 // constraint) 1474 // FIXME this rounding up should not be done if no HAL SRC 1475 uint32_t truncMult = (uint32_t) multiplier; 1476 if ((truncMult & 1)) { 1477 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1478 ++truncMult; 1479 } 1480 } 1481 multiplier = (double) truncMult; 1482 } 1483 } 1484 mNormalFrameCount = multiplier * mFrameCount; 1485 // round up to nearest 16 frames to satisfy AudioMixer 1486 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1487 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1488 mNormalFrameCount); 1489 1490 delete[] mMixBuffer; 1491 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1492 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1493 1494 // force reconfiguration of effect chains and engines to take new buffer size and audio 1495 // parameters into account 1496 // Note that mLock is not held when readOutputParameters() is called from the constructor 1497 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1498 // matter. 1499 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1500 Vector< sp<EffectChain> > effectChains = mEffectChains; 1501 for (size_t i = 0; i < effectChains.size(); i ++) { 1502 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1503 } 1504} 1505 1506 1507status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1508{ 1509 if (halFrames == NULL || dspFrames == NULL) { 1510 return BAD_VALUE; 1511 } 1512 Mutex::Autolock _l(mLock); 1513 if (initCheck() != NO_ERROR) { 1514 return INVALID_OPERATION; 1515 } 1516 size_t framesWritten = mBytesWritten / mFrameSize; 1517 *halFrames = framesWritten; 1518 1519 if (isSuspended()) { 1520 // return an estimation of rendered frames when the output is suspended 1521 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1522 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1523 return NO_ERROR; 1524 } else { 1525 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1526 } 1527} 1528 1529uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1530{ 1531 Mutex::Autolock _l(mLock); 1532 uint32_t result = 0; 1533 if (getEffectChain_l(sessionId) != 0) { 1534 result = EFFECT_SESSION; 1535 } 1536 1537 for (size_t i = 0; i < mTracks.size(); ++i) { 1538 sp<Track> track = mTracks[i]; 1539 if (sessionId == track->sessionId() && !track->isInvalid()) { 1540 result |= TRACK_SESSION; 1541 break; 1542 } 1543 } 1544 1545 return result; 1546} 1547 1548uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1549{ 1550 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1551 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1552 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1553 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1554 } 1555 for (size_t i = 0; i < mTracks.size(); i++) { 1556 sp<Track> track = mTracks[i]; 1557 if (sessionId == track->sessionId() && !track->isInvalid()) { 1558 return AudioSystem::getStrategyForStream(track->streamType()); 1559 } 1560 } 1561 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1562} 1563 1564 1565AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1566{ 1567 Mutex::Autolock _l(mLock); 1568 return mOutput; 1569} 1570 1571AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1572{ 1573 Mutex::Autolock _l(mLock); 1574 AudioStreamOut *output = mOutput; 1575 mOutput = NULL; 1576 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1577 // must push a NULL and wait for ack 1578 mOutputSink.clear(); 1579 mPipeSink.clear(); 1580 mNormalSink.clear(); 1581 return output; 1582} 1583 1584// this method must always be called either with ThreadBase mLock held or inside the thread loop 1585audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1586{ 1587 if (mOutput == NULL) { 1588 return NULL; 1589 } 1590 return &mOutput->stream->common; 1591} 1592 1593uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1594{ 1595 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1596} 1597 1598status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1599{ 1600 if (!isValidSyncEvent(event)) { 1601 return BAD_VALUE; 1602 } 1603 1604 Mutex::Autolock _l(mLock); 1605 1606 for (size_t i = 0; i < mTracks.size(); ++i) { 1607 sp<Track> track = mTracks[i]; 1608 if (event->triggerSession() == track->sessionId()) { 1609 (void) track->setSyncEvent(event); 1610 return NO_ERROR; 1611 } 1612 } 1613 1614 return NAME_NOT_FOUND; 1615} 1616 1617bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1618{ 1619 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1620} 1621 1622void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1623 const Vector< sp<Track> >& tracksToRemove) 1624{ 1625 size_t count = tracksToRemove.size(); 1626 if (CC_UNLIKELY(count)) { 1627 for (size_t i = 0 ; i < count ; i++) { 1628 const sp<Track>& track = tracksToRemove.itemAt(i); 1629 if ((track->sharedBuffer() != 0) && 1630 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1631 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1632 } 1633 } 1634 } 1635 1636} 1637 1638void AudioFlinger::PlaybackThread::checkSilentMode_l() 1639{ 1640 if (!mMasterMute) { 1641 char value[PROPERTY_VALUE_MAX]; 1642 if (property_get("ro.audio.silent", value, "0") > 0) { 1643 char *endptr; 1644 unsigned long ul = strtoul(value, &endptr, 0); 1645 if (*endptr == '\0' && ul != 0) { 1646 ALOGD("Silence is golden"); 1647 // The setprop command will not allow a property to be changed after 1648 // the first time it is set, so we don't have to worry about un-muting. 1649 setMasterMute_l(true); 1650 } 1651 } 1652 } 1653} 1654 1655// shared by MIXER and DIRECT, overridden by DUPLICATING 1656void AudioFlinger::PlaybackThread::threadLoop_write() 1657{ 1658 // FIXME rewrite to reduce number of system calls 1659 mLastWriteTime = systemTime(); 1660 mInWrite = true; 1661 int bytesWritten; 1662 1663 // If an NBAIO sink is present, use it to write the normal mixer's submix 1664 if (mNormalSink != 0) { 1665#define mBitShift 2 // FIXME 1666 size_t count = mixBufferSize >> mBitShift; 1667 ATRACE_BEGIN("write"); 1668 // update the setpoint when AudioFlinger::mScreenState changes 1669 uint32_t screenState = AudioFlinger::mScreenState; 1670 if (screenState != mScreenState) { 1671 mScreenState = screenState; 1672 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1673 if (pipe != NULL) { 1674 pipe->setAvgFrames((mScreenState & 1) ? 1675 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1676 } 1677 } 1678 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1679 ATRACE_END(); 1680 if (framesWritten > 0) { 1681 bytesWritten = framesWritten << mBitShift; 1682 } else { 1683 bytesWritten = framesWritten; 1684 } 1685 // otherwise use the HAL / AudioStreamOut directly 1686 } else { 1687 // Direct output thread. 1688 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1689 } 1690 1691 if (bytesWritten > 0) { 1692 mBytesWritten += mixBufferSize; 1693 } 1694 mNumWrites++; 1695 mInWrite = false; 1696} 1697 1698/* 1699The derived values that are cached: 1700 - mixBufferSize from frame count * frame size 1701 - activeSleepTime from activeSleepTimeUs() 1702 - idleSleepTime from idleSleepTimeUs() 1703 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1704 - maxPeriod from frame count and sample rate (MIXER only) 1705 1706The parameters that affect these derived values are: 1707 - frame count 1708 - frame size 1709 - sample rate 1710 - device type: A2DP or not 1711 - device latency 1712 - format: PCM or not 1713 - active sleep time 1714 - idle sleep time 1715*/ 1716 1717void AudioFlinger::PlaybackThread::cacheParameters_l() 1718{ 1719 mixBufferSize = mNormalFrameCount * mFrameSize; 1720 activeSleepTime = activeSleepTimeUs(); 1721 idleSleepTime = idleSleepTimeUs(); 1722} 1723 1724void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1725{ 1726 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1727 this, streamType, mTracks.size()); 1728 Mutex::Autolock _l(mLock); 1729 1730 size_t size = mTracks.size(); 1731 for (size_t i = 0; i < size; i++) { 1732 sp<Track> t = mTracks[i]; 1733 if (t->streamType() == streamType) { 1734 t->invalidate(); 1735 } 1736 } 1737} 1738 1739status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1740{ 1741 int session = chain->sessionId(); 1742 int16_t *buffer = mMixBuffer; 1743 bool ownsBuffer = false; 1744 1745 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1746 if (session > 0) { 1747 // Only one effect chain can be present in direct output thread and it uses 1748 // the mix buffer as input 1749 if (mType != DIRECT) { 1750 size_t numSamples = mNormalFrameCount * mChannelCount; 1751 buffer = new int16_t[numSamples]; 1752 memset(buffer, 0, numSamples * sizeof(int16_t)); 1753 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1754 ownsBuffer = true; 1755 } 1756 1757 // Attach all tracks with same session ID to this chain. 1758 for (size_t i = 0; i < mTracks.size(); ++i) { 1759 sp<Track> track = mTracks[i]; 1760 if (session == track->sessionId()) { 1761 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1762 buffer); 1763 track->setMainBuffer(buffer); 1764 chain->incTrackCnt(); 1765 } 1766 } 1767 1768 // indicate all active tracks in the chain 1769 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1770 sp<Track> track = mActiveTracks[i].promote(); 1771 if (track == 0) { 1772 continue; 1773 } 1774 if (session == track->sessionId()) { 1775 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1776 chain->incActiveTrackCnt(); 1777 } 1778 } 1779 } 1780 1781 chain->setInBuffer(buffer, ownsBuffer); 1782 chain->setOutBuffer(mMixBuffer); 1783 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1784 // chains list in order to be processed last as it contains output stage effects 1785 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1786 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1787 // after track specific effects and before output stage 1788 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1789 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1790 // Effect chain for other sessions are inserted at beginning of effect 1791 // chains list to be processed before output mix effects. Relative order between other 1792 // sessions is not important 1793 size_t size = mEffectChains.size(); 1794 size_t i = 0; 1795 for (i = 0; i < size; i++) { 1796 if (mEffectChains[i]->sessionId() < session) { 1797 break; 1798 } 1799 } 1800 mEffectChains.insertAt(chain, i); 1801 checkSuspendOnAddEffectChain_l(chain); 1802 1803 return NO_ERROR; 1804} 1805 1806size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1807{ 1808 int session = chain->sessionId(); 1809 1810 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1811 1812 for (size_t i = 0; i < mEffectChains.size(); i++) { 1813 if (chain == mEffectChains[i]) { 1814 mEffectChains.removeAt(i); 1815 // detach all active tracks from the chain 1816 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1817 sp<Track> track = mActiveTracks[i].promote(); 1818 if (track == 0) { 1819 continue; 1820 } 1821 if (session == track->sessionId()) { 1822 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1823 chain.get(), session); 1824 chain->decActiveTrackCnt(); 1825 } 1826 } 1827 1828 // detach all tracks with same session ID from this chain 1829 for (size_t i = 0; i < mTracks.size(); ++i) { 1830 sp<Track> track = mTracks[i]; 1831 if (session == track->sessionId()) { 1832 track->setMainBuffer(mMixBuffer); 1833 chain->decTrackCnt(); 1834 } 1835 } 1836 break; 1837 } 1838 } 1839 return mEffectChains.size(); 1840} 1841 1842status_t AudioFlinger::PlaybackThread::attachAuxEffect( 1843 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1844{ 1845 Mutex::Autolock _l(mLock); 1846 return attachAuxEffect_l(track, EffectId); 1847} 1848 1849status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 1850 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1851{ 1852 status_t status = NO_ERROR; 1853 1854 if (EffectId == 0) { 1855 track->setAuxBuffer(0, NULL); 1856 } else { 1857 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 1858 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 1859 if (effect != 0) { 1860 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1861 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 1862 } else { 1863 status = INVALID_OPERATION; 1864 } 1865 } else { 1866 status = BAD_VALUE; 1867 } 1868 } 1869 return status; 1870} 1871 1872void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 1873{ 1874 for (size_t i = 0; i < mTracks.size(); ++i) { 1875 sp<Track> track = mTracks[i]; 1876 if (track->auxEffectId() == effectId) { 1877 attachAuxEffect_l(track, 0); 1878 } 1879 } 1880} 1881 1882bool AudioFlinger::PlaybackThread::threadLoop() 1883{ 1884 Vector< sp<Track> > tracksToRemove; 1885 1886 standbyTime = systemTime(); 1887 1888 // MIXER 1889 nsecs_t lastWarning = 0; 1890 1891 // DUPLICATING 1892 // FIXME could this be made local to while loop? 1893 writeFrames = 0; 1894 1895 cacheParameters_l(); 1896 sleepTime = idleSleepTime; 1897 1898 if (mType == MIXER) { 1899 sleepTimeShift = 0; 1900 } 1901 1902 CpuStats cpuStats; 1903 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 1904 1905 acquireWakeLock(); 1906 1907 // mNBLogWriter->log can only be called while thread mutex mLock is held. 1908 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 1909 // and then that string will be logged at the next convenient opportunity. 1910 const char *logString = NULL; 1911 1912 while (!exitPending()) 1913 { 1914 cpuStats.sample(myName); 1915 1916 Vector< sp<EffectChain> > effectChains; 1917 1918 processConfigEvents(); 1919 1920 { // scope for mLock 1921 1922 Mutex::Autolock _l(mLock); 1923 1924 if (logString != NULL) { 1925 mNBLogWriter->logTimestamp(); 1926 mNBLogWriter->log(logString); 1927 logString = NULL; 1928 } 1929 1930 if (checkForNewParameters_l()) { 1931 cacheParameters_l(); 1932 } 1933 1934 saveOutputTracks(); 1935 1936 // put audio hardware into standby after short delay 1937 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1938 isSuspended())) { 1939 if (!mStandby) { 1940 1941 threadLoop_standby(); 1942 1943 mNBLogWriter->log("standby"); 1944 mStandby = true; 1945 } 1946 1947 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1948 // we're about to wait, flush the binder command buffer 1949 IPCThreadState::self()->flushCommands(); 1950 1951 clearOutputTracks(); 1952 1953 if (exitPending()) { 1954 break; 1955 } 1956 1957 releaseWakeLock_l(); 1958 // wait until we have something to do... 1959 ALOGV("%s going to sleep", myName.string()); 1960 mWaitWorkCV.wait(mLock); 1961 ALOGV("%s waking up", myName.string()); 1962 acquireWakeLock_l(); 1963 1964 mMixerStatus = MIXER_IDLE; 1965 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1966 mBytesWritten = 0; 1967 1968 checkSilentMode_l(); 1969 1970 standbyTime = systemTime() + standbyDelay; 1971 sleepTime = idleSleepTime; 1972 if (mType == MIXER) { 1973 sleepTimeShift = 0; 1974 } 1975 1976 continue; 1977 } 1978 } 1979 1980 // mMixerStatusIgnoringFastTracks is also updated internally 1981 mMixerStatus = prepareTracks_l(&tracksToRemove); 1982 1983 // prevent any changes in effect chain list and in each effect chain 1984 // during mixing and effect process as the audio buffers could be deleted 1985 // or modified if an effect is created or deleted 1986 lockEffectChains_l(effectChains); 1987 } 1988 1989 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 1990 threadLoop_mix(); 1991 } else { 1992 threadLoop_sleepTime(); 1993 } 1994 1995 if (isSuspended()) { 1996 sleepTime = suspendSleepTimeUs(); 1997 mBytesWritten += mixBufferSize; 1998 } 1999 2000 // only process effects if we're going to write 2001 if (sleepTime == 0) { 2002 for (size_t i = 0; i < effectChains.size(); i ++) { 2003 effectChains[i]->process_l(); 2004 } 2005 } 2006 2007 // enable changes in effect chain 2008 unlockEffectChains(effectChains); 2009 2010 // sleepTime == 0 means we must write to audio hardware 2011 if (sleepTime == 0) { 2012 2013 threadLoop_write(); 2014 2015if (mType == MIXER) { 2016 // write blocked detection 2017 nsecs_t now = systemTime(); 2018 nsecs_t delta = now - mLastWriteTime; 2019 if (!mStandby && delta > maxPeriod) { 2020 mNumDelayedWrites++; 2021 if ((now - lastWarning) > kWarningThrottleNs) { 2022 ATRACE_NAME("underrun"); 2023 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2024 ns2ms(delta), mNumDelayedWrites, this); 2025 lastWarning = now; 2026 } 2027 } 2028} 2029 2030 mStandby = false; 2031 } else { 2032 usleep(sleepTime); 2033 } 2034 2035 // Finally let go of removed track(s), without the lock held 2036 // since we can't guarantee the destructors won't acquire that 2037 // same lock. This will also mutate and push a new fast mixer state. 2038 threadLoop_removeTracks(tracksToRemove); 2039 if (tracksToRemove.size() > 0) { 2040 logString = "remove"; 2041 } 2042 tracksToRemove.clear(); 2043 2044 // FIXME I don't understand the need for this here; 2045 // it was in the original code but maybe the 2046 // assignment in saveOutputTracks() makes this unnecessary? 2047 clearOutputTracks(); 2048 2049 // Effect chains will be actually deleted here if they were removed from 2050 // mEffectChains list during mixing or effects processing 2051 effectChains.clear(); 2052 2053 // FIXME Note that the above .clear() is no longer necessary since effectChains 2054 // is now local to this block, but will keep it for now (at least until merge done). 2055 } 2056 2057 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2058 if (mType == MIXER || mType == DIRECT) { 2059 // put output stream into standby mode 2060 if (!mStandby) { 2061 mOutput->stream->common.standby(&mOutput->stream->common); 2062 } 2063 } 2064 2065 releaseWakeLock(); 2066 2067 ALOGV("Thread %p type %d exiting", this, mType); 2068 return false; 2069} 2070 2071 2072// ---------------------------------------------------------------------------- 2073 2074AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2075 audio_io_handle_t id, audio_devices_t device, type_t type) 2076 : PlaybackThread(audioFlinger, output, id, device, type), 2077 // mAudioMixer below 2078 // mFastMixer below 2079 mFastMixerFutex(0) 2080 // mOutputSink below 2081 // mPipeSink below 2082 // mNormalSink below 2083{ 2084 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2085 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2086 "mFrameCount=%d, mNormalFrameCount=%d", 2087 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2088 mNormalFrameCount); 2089 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2090 2091 // FIXME - Current mixer implementation only supports stereo output 2092 if (mChannelCount != FCC_2) { 2093 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2094 } 2095 2096 // create an NBAIO sink for the HAL output stream, and negotiate 2097 mOutputSink = new AudioStreamOutSink(output->stream); 2098 size_t numCounterOffers = 0; 2099 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2100 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2101 ALOG_ASSERT(index == 0); 2102 2103 // initialize fast mixer depending on configuration 2104 bool initFastMixer; 2105 switch (kUseFastMixer) { 2106 case FastMixer_Never: 2107 initFastMixer = false; 2108 break; 2109 case FastMixer_Always: 2110 initFastMixer = true; 2111 break; 2112 case FastMixer_Static: 2113 case FastMixer_Dynamic: 2114 initFastMixer = mFrameCount < mNormalFrameCount; 2115 break; 2116 } 2117 if (initFastMixer) { 2118 2119 // create a MonoPipe to connect our submix to FastMixer 2120 NBAIO_Format format = mOutputSink->format(); 2121 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2122 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2123 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2124 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2125 const NBAIO_Format offers[1] = {format}; 2126 size_t numCounterOffers = 0; 2127 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2128 ALOG_ASSERT(index == 0); 2129 monoPipe->setAvgFrames((mScreenState & 1) ? 2130 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2131 mPipeSink = monoPipe; 2132 2133#ifdef TEE_SINK_FRAMES 2134 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2135 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2136 numCounterOffers = 0; 2137 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2138 ALOG_ASSERT(index == 0); 2139 mTeeSink = teeSink; 2140 PipeReader *teeSource = new PipeReader(*teeSink); 2141 numCounterOffers = 0; 2142 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2143 ALOG_ASSERT(index == 0); 2144 mTeeSource = teeSource; 2145#endif 2146 2147 // create fast mixer and configure it initially with just one fast track for our submix 2148 mFastMixer = new FastMixer(); 2149 FastMixerStateQueue *sq = mFastMixer->sq(); 2150#ifdef STATE_QUEUE_DUMP 2151 sq->setObserverDump(&mStateQueueObserverDump); 2152 sq->setMutatorDump(&mStateQueueMutatorDump); 2153#endif 2154 FastMixerState *state = sq->begin(); 2155 FastTrack *fastTrack = &state->mFastTracks[0]; 2156 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2157 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2158 mNBLogWriter->logf("fastTrack0 bp=%p", fastTrack->mBufferProvider); 2159 fastTrack->mVolumeProvider = NULL; 2160 fastTrack->mGeneration++; 2161 state->mFastTracksGen++; 2162 state->mTrackMask = 1; 2163 // fast mixer will use the HAL output sink 2164 state->mOutputSink = mOutputSink.get(); 2165 state->mOutputSinkGen++; 2166 state->mFrameCount = mFrameCount; 2167 state->mCommand = FastMixerState::COLD_IDLE; 2168 // already done in constructor initialization list 2169 //mFastMixerFutex = 0; 2170 state->mColdFutexAddr = &mFastMixerFutex; 2171 state->mColdGen++; 2172 state->mDumpState = &mFastMixerDumpState; 2173 state->mTeeSink = mTeeSink.get(); 2174 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2175 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2176 sq->end(); 2177 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2178 2179 // start the fast mixer 2180 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2181 pid_t tid = mFastMixer->getTid(); 2182 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2183 if (err != 0) { 2184 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2185 kPriorityFastMixer, getpid_cached, tid, err); 2186 } 2187 2188#ifdef AUDIO_WATCHDOG 2189 // create and start the watchdog 2190 mAudioWatchdog = new AudioWatchdog(); 2191 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2192 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2193 tid = mAudioWatchdog->getTid(); 2194 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2195 if (err != 0) { 2196 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2197 kPriorityFastMixer, getpid_cached, tid, err); 2198 } 2199#endif 2200 2201 } else { 2202 mFastMixer = NULL; 2203 } 2204 2205 switch (kUseFastMixer) { 2206 case FastMixer_Never: 2207 case FastMixer_Dynamic: 2208 mNormalSink = mOutputSink; 2209 break; 2210 case FastMixer_Always: 2211 mNormalSink = mPipeSink; 2212 break; 2213 case FastMixer_Static: 2214 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2215 break; 2216 } 2217} 2218 2219AudioFlinger::MixerThread::~MixerThread() 2220{ 2221 if (mFastMixer != NULL) { 2222 FastMixerStateQueue *sq = mFastMixer->sq(); 2223 FastMixerState *state = sq->begin(); 2224 if (state->mCommand == FastMixerState::COLD_IDLE) { 2225 int32_t old = android_atomic_inc(&mFastMixerFutex); 2226 if (old == -1) { 2227 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2228 } 2229 } 2230 state->mCommand = FastMixerState::EXIT; 2231 sq->end(); 2232 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2233 mFastMixer->join(); 2234 // Though the fast mixer thread has exited, it's state queue is still valid. 2235 // We'll use that extract the final state which contains one remaining fast track 2236 // corresponding to our sub-mix. 2237 state = sq->begin(); 2238 ALOG_ASSERT(state->mTrackMask == 1); 2239 FastTrack *fastTrack = &state->mFastTracks[0]; 2240 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2241 delete fastTrack->mBufferProvider; 2242 sq->end(false /*didModify*/); 2243 delete mFastMixer; 2244#ifdef AUDIO_WATCHDOG 2245 if (mAudioWatchdog != 0) { 2246 mAudioWatchdog->requestExit(); 2247 mAudioWatchdog->requestExitAndWait(); 2248 mAudioWatchdog.clear(); 2249 } 2250#endif 2251 } 2252 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2253 delete mAudioMixer; 2254} 2255 2256 2257uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2258{ 2259 if (mFastMixer != NULL) { 2260 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2261 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2262 } 2263 return latency; 2264} 2265 2266 2267void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2268{ 2269 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2270} 2271 2272void AudioFlinger::MixerThread::threadLoop_write() 2273{ 2274 // FIXME we should only do one push per cycle; confirm this is true 2275 // Start the fast mixer if it's not already running 2276 if (mFastMixer != NULL) { 2277 FastMixerStateQueue *sq = mFastMixer->sq(); 2278 FastMixerState *state = sq->begin(); 2279 if (state->mCommand != FastMixerState::MIX_WRITE && 2280 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2281 if (state->mCommand == FastMixerState::COLD_IDLE) { 2282 int32_t old = android_atomic_inc(&mFastMixerFutex); 2283 if (old == -1) { 2284 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2285 } 2286#ifdef AUDIO_WATCHDOG 2287 if (mAudioWatchdog != 0) { 2288 mAudioWatchdog->resume(); 2289 } 2290#endif 2291 } 2292 state->mCommand = FastMixerState::MIX_WRITE; 2293 sq->end(); 2294 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2295 if (kUseFastMixer == FastMixer_Dynamic) { 2296 mNormalSink = mPipeSink; 2297 } 2298 } else { 2299 sq->end(false /*didModify*/); 2300 } 2301 } 2302 PlaybackThread::threadLoop_write(); 2303} 2304 2305void AudioFlinger::MixerThread::threadLoop_standby() 2306{ 2307 // Idle the fast mixer if it's currently running 2308 if (mFastMixer != NULL) { 2309 FastMixerStateQueue *sq = mFastMixer->sq(); 2310 FastMixerState *state = sq->begin(); 2311 if (!(state->mCommand & FastMixerState::IDLE)) { 2312 state->mCommand = FastMixerState::COLD_IDLE; 2313 state->mColdFutexAddr = &mFastMixerFutex; 2314 state->mColdGen++; 2315 mFastMixerFutex = 0; 2316 sq->end(); 2317 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2318 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2319 if (kUseFastMixer == FastMixer_Dynamic) { 2320 mNormalSink = mOutputSink; 2321 } 2322#ifdef AUDIO_WATCHDOG 2323 if (mAudioWatchdog != 0) { 2324 mAudioWatchdog->pause(); 2325 } 2326#endif 2327 } else { 2328 sq->end(false /*didModify*/); 2329 } 2330 } 2331 PlaybackThread::threadLoop_standby(); 2332} 2333 2334// shared by MIXER and DIRECT, overridden by DUPLICATING 2335void AudioFlinger::PlaybackThread::threadLoop_standby() 2336{ 2337 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2338 mOutput->stream->common.standby(&mOutput->stream->common); 2339} 2340 2341void AudioFlinger::MixerThread::threadLoop_mix() 2342{ 2343 // obtain the presentation timestamp of the next output buffer 2344 int64_t pts; 2345 status_t status = INVALID_OPERATION; 2346 2347 if (mNormalSink != 0) { 2348 status = mNormalSink->getNextWriteTimestamp(&pts); 2349 } else { 2350 status = mOutputSink->getNextWriteTimestamp(&pts); 2351 } 2352 2353 if (status != NO_ERROR) { 2354 pts = AudioBufferProvider::kInvalidPTS; 2355 } 2356 2357 // mix buffers... 2358 mAudioMixer->process(pts); 2359 // increase sleep time progressively when application underrun condition clears. 2360 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2361 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2362 // such that we would underrun the audio HAL. 2363 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2364 sleepTimeShift--; 2365 } 2366 sleepTime = 0; 2367 standbyTime = systemTime() + standbyDelay; 2368 //TODO: delay standby when effects have a tail 2369} 2370 2371void AudioFlinger::MixerThread::threadLoop_sleepTime() 2372{ 2373 // If no tracks are ready, sleep once for the duration of an output 2374 // buffer size, then write 0s to the output 2375 if (sleepTime == 0) { 2376 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2377 sleepTime = activeSleepTime >> sleepTimeShift; 2378 if (sleepTime < kMinThreadSleepTimeUs) { 2379 sleepTime = kMinThreadSleepTimeUs; 2380 } 2381 // reduce sleep time in case of consecutive application underruns to avoid 2382 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2383 // duration we would end up writing less data than needed by the audio HAL if 2384 // the condition persists. 2385 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2386 sleepTimeShift++; 2387 } 2388 } else { 2389 sleepTime = idleSleepTime; 2390 } 2391 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2392 memset (mMixBuffer, 0, mixBufferSize); 2393 sleepTime = 0; 2394 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2395 "anticipated start"); 2396 } 2397 // TODO add standby time extension fct of effect tail 2398} 2399 2400// prepareTracks_l() must be called with ThreadBase::mLock held 2401AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2402 Vector< sp<Track> > *tracksToRemove) 2403{ 2404 2405 mixer_state mixerStatus = MIXER_IDLE; 2406 // find out which tracks need to be processed 2407 size_t count = mActiveTracks.size(); 2408 size_t mixedTracks = 0; 2409 size_t tracksWithEffect = 0; 2410 // counts only _active_ fast tracks 2411 size_t fastTracks = 0; 2412 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2413 2414 float masterVolume = mMasterVolume; 2415 bool masterMute = mMasterMute; 2416 2417 if (masterMute) { 2418 masterVolume = 0; 2419 } 2420 // Delegate master volume control to effect in output mix effect chain if needed 2421 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2422 if (chain != 0) { 2423 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2424 chain->setVolume_l(&v, &v); 2425 masterVolume = (float)((v + (1 << 23)) >> 24); 2426 chain.clear(); 2427 } 2428 2429 // prepare a new state to push 2430 FastMixerStateQueue *sq = NULL; 2431 FastMixerState *state = NULL; 2432 bool didModify = false; 2433 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2434 if (mFastMixer != NULL) { 2435 sq = mFastMixer->sq(); 2436 state = sq->begin(); 2437 } 2438 2439 for (size_t i=0 ; i<count ; i++) { 2440 sp<Track> t = mActiveTracks[i].promote(); 2441 if (t == 0) { 2442 continue; 2443 } 2444 2445 // this const just means the local variable doesn't change 2446 Track* const track = t.get(); 2447 2448 // process fast tracks 2449 if (track->isFastTrack()) { 2450 2451 // It's theoretically possible (though unlikely) for a fast track to be created 2452 // and then removed within the same normal mix cycle. This is not a problem, as 2453 // the track never becomes active so it's fast mixer slot is never touched. 2454 // The converse, of removing an (active) track and then creating a new track 2455 // at the identical fast mixer slot within the same normal mix cycle, 2456 // is impossible because the slot isn't marked available until the end of each cycle. 2457 int j = track->mFastIndex; 2458 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2459 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2460 FastTrack *fastTrack = &state->mFastTracks[j]; 2461 2462 // Determine whether the track is currently in underrun condition, 2463 // and whether it had a recent underrun. 2464 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2465 FastTrackUnderruns underruns = ftDump->mUnderruns; 2466 uint32_t recentFull = (underruns.mBitFields.mFull - 2467 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2468 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2469 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2470 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2471 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2472 uint32_t recentUnderruns = recentPartial + recentEmpty; 2473 track->mObservedUnderruns = underruns; 2474 // don't count underruns that occur while stopping or pausing 2475 // or stopped which can occur when flush() is called while active 2476 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2477 track->mUnderrunCount += recentUnderruns; 2478 } 2479 2480 // This is similar to the state machine for normal tracks, 2481 // with a few modifications for fast tracks. 2482 bool isActive = true; 2483 switch (track->mState) { 2484 case TrackBase::STOPPING_1: 2485 // track stays active in STOPPING_1 state until first underrun 2486 if (recentUnderruns > 0) { 2487 track->mState = TrackBase::STOPPING_2; 2488 } 2489 break; 2490 case TrackBase::PAUSING: 2491 // ramp down is not yet implemented 2492 track->setPaused(); 2493 break; 2494 case TrackBase::RESUMING: 2495 // ramp up is not yet implemented 2496 track->mState = TrackBase::ACTIVE; 2497 break; 2498 case TrackBase::ACTIVE: 2499 if (recentFull > 0 || recentPartial > 0) { 2500 // track has provided at least some frames recently: reset retry count 2501 track->mRetryCount = kMaxTrackRetries; 2502 } 2503 if (recentUnderruns == 0) { 2504 // no recent underruns: stay active 2505 break; 2506 } 2507 // there has recently been an underrun of some kind 2508 if (track->sharedBuffer() == 0) { 2509 // were any of the recent underruns "empty" (no frames available)? 2510 if (recentEmpty == 0) { 2511 // no, then ignore the partial underruns as they are allowed indefinitely 2512 break; 2513 } 2514 // there has recently been an "empty" underrun: decrement the retry counter 2515 if (--(track->mRetryCount) > 0) { 2516 break; 2517 } 2518 // indicate to client process that the track was disabled because of underrun; 2519 // it will then automatically call start() when data is available 2520 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2521 // remove from active list, but state remains ACTIVE [confusing but true] 2522 isActive = false; 2523 break; 2524 } 2525 // fall through 2526 case TrackBase::STOPPING_2: 2527 case TrackBase::PAUSED: 2528 case TrackBase::TERMINATED: 2529 case TrackBase::STOPPED: 2530 case TrackBase::FLUSHED: // flush() while active 2531 // Check for presentation complete if track is inactive 2532 // We have consumed all the buffers of this track. 2533 // This would be incomplete if we auto-paused on underrun 2534 { 2535 size_t audioHALFrames = 2536 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2537 size_t framesWritten = mBytesWritten / mFrameSize; 2538 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2539 // track stays in active list until presentation is complete 2540 break; 2541 } 2542 } 2543 if (track->isStopping_2()) { 2544 track->mState = TrackBase::STOPPED; 2545 } 2546 if (track->isStopped()) { 2547 // Can't reset directly, as fast mixer is still polling this track 2548 // track->reset(); 2549 // So instead mark this track as needing to be reset after push with ack 2550 resetMask |= 1 << i; 2551 } 2552 isActive = false; 2553 break; 2554 case TrackBase::IDLE: 2555 default: 2556 LOG_FATAL("unexpected track state %d", track->mState); 2557 } 2558 2559 if (isActive) { 2560 // was it previously inactive? 2561 if (!(state->mTrackMask & (1 << j))) { 2562 ExtendedAudioBufferProvider *eabp = track; 2563 mNBLogWriter->logf("fastTrack j=%d bp=%p", j, eabp); 2564 VolumeProvider *vp = track; 2565 fastTrack->mBufferProvider = eabp; 2566 fastTrack->mVolumeProvider = vp; 2567 fastTrack->mSampleRate = track->mSampleRate; 2568 fastTrack->mChannelMask = track->mChannelMask; 2569 fastTrack->mGeneration++; 2570 state->mTrackMask |= 1 << j; 2571 didModify = true; 2572 // no acknowledgement required for newly active tracks 2573 } 2574 // cache the combined master volume and stream type volume for fast mixer; this 2575 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2576 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2577 ++fastTracks; 2578 } else { 2579 // was it previously active? 2580 if (state->mTrackMask & (1 << j)) { 2581 fastTrack->mBufferProvider = NULL; 2582 fastTrack->mGeneration++; 2583 state->mTrackMask &= ~(1 << j); 2584 didModify = true; 2585 // If any fast tracks were removed, we must wait for acknowledgement 2586 // because we're about to decrement the last sp<> on those tracks. 2587 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2588 } else { 2589 LOG_FATAL("fast track %d should have been active", j); 2590 } 2591 tracksToRemove->add(track); 2592 // Avoids a misleading display in dumpsys 2593 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2594 } 2595 continue; 2596 } 2597 2598 { // local variable scope to avoid goto warning 2599 2600 audio_track_cblk_t* cblk = track->cblk(); 2601 2602 // The first time a track is added we wait 2603 // for all its buffers to be filled before processing it 2604 int name = track->name(); 2605 // make sure that we have enough frames to mix one full buffer. 2606 // enforce this condition only once to enable draining the buffer in case the client 2607 // app does not call stop() and relies on underrun to stop: 2608 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2609 // during last round 2610 uint32_t minFrames = 1; 2611 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2612 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2613 if (t->sampleRate() == mSampleRate) { 2614 minFrames = mNormalFrameCount; 2615 } else { 2616 // +1 for rounding and +1 for additional sample needed for interpolation 2617 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2618 // add frames already consumed but not yet released by the resampler 2619 // because cblk->framesReady() will include these frames 2620 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2621 // the minimum track buffer size is normally twice the number of frames necessary 2622 // to fill one buffer and the resampler should not leave more than one buffer worth 2623 // of unreleased frames after each pass, but just in case... 2624 ALOG_ASSERT(minFrames <= cblk->frameCount_); 2625 } 2626 } 2627 if ((track->framesReady() >= minFrames) && track->isReady() && 2628 !track->isPaused() && !track->isTerminated()) 2629 { 2630 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2631 this); 2632 2633 mixedTracks++; 2634 2635 // track->mainBuffer() != mMixBuffer means there is an effect chain 2636 // connected to the track 2637 chain.clear(); 2638 if (track->mainBuffer() != mMixBuffer) { 2639 chain = getEffectChain_l(track->sessionId()); 2640 // Delegate volume control to effect in track effect chain if needed 2641 if (chain != 0) { 2642 tracksWithEffect++; 2643 } else { 2644 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2645 "session %d", 2646 name, track->sessionId()); 2647 } 2648 } 2649 2650 2651 int param = AudioMixer::VOLUME; 2652 if (track->mFillingUpStatus == Track::FS_FILLED) { 2653 // no ramp for the first volume setting 2654 track->mFillingUpStatus = Track::FS_ACTIVE; 2655 if (track->mState == TrackBase::RESUMING) { 2656 track->mState = TrackBase::ACTIVE; 2657 param = AudioMixer::RAMP_VOLUME; 2658 } 2659 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2660 } else if (cblk->server != 0) { 2661 // If the track is stopped before the first frame was mixed, 2662 // do not apply ramp 2663 param = AudioMixer::RAMP_VOLUME; 2664 } 2665 2666 // compute volume for this track 2667 uint32_t vl, vr, va; 2668 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2669 vl = vr = va = 0; 2670 if (track->isPausing()) { 2671 track->setPaused(); 2672 } 2673 } else { 2674 2675 // read original volumes with volume control 2676 float typeVolume = mStreamTypes[track->streamType()].volume; 2677 float v = masterVolume * typeVolume; 2678 ServerProxy *proxy = track->mServerProxy; 2679 uint32_t vlr = proxy->getVolumeLR(); 2680 vl = vlr & 0xFFFF; 2681 vr = vlr >> 16; 2682 // track volumes come from shared memory, so can't be trusted and must be clamped 2683 if (vl > MAX_GAIN_INT) { 2684 ALOGV("Track left volume out of range: %04X", vl); 2685 vl = MAX_GAIN_INT; 2686 } 2687 if (vr > MAX_GAIN_INT) { 2688 ALOGV("Track right volume out of range: %04X", vr); 2689 vr = MAX_GAIN_INT; 2690 } 2691 // now apply the master volume and stream type volume 2692 vl = (uint32_t)(v * vl) << 12; 2693 vr = (uint32_t)(v * vr) << 12; 2694 // assuming master volume and stream type volume each go up to 1.0, 2695 // vl and vr are now in 8.24 format 2696 2697 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2698 // send level comes from shared memory and so may be corrupt 2699 if (sendLevel > MAX_GAIN_INT) { 2700 ALOGV("Track send level out of range: %04X", sendLevel); 2701 sendLevel = MAX_GAIN_INT; 2702 } 2703 va = (uint32_t)(v * sendLevel); 2704 } 2705 // Delegate volume control to effect in track effect chain if needed 2706 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2707 // Do not ramp volume if volume is controlled by effect 2708 param = AudioMixer::VOLUME; 2709 track->mHasVolumeController = true; 2710 } else { 2711 // force no volume ramp when volume controller was just disabled or removed 2712 // from effect chain to avoid volume spike 2713 if (track->mHasVolumeController) { 2714 param = AudioMixer::VOLUME; 2715 } 2716 track->mHasVolumeController = false; 2717 } 2718 2719 // Convert volumes from 8.24 to 4.12 format 2720 // This additional clamping is needed in case chain->setVolume_l() overshot 2721 vl = (vl + (1 << 11)) >> 12; 2722 if (vl > MAX_GAIN_INT) { 2723 vl = MAX_GAIN_INT; 2724 } 2725 vr = (vr + (1 << 11)) >> 12; 2726 if (vr > MAX_GAIN_INT) { 2727 vr = MAX_GAIN_INT; 2728 } 2729 2730 if (va > MAX_GAIN_INT) { 2731 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2732 } 2733 2734 // XXX: these things DON'T need to be done each time 2735 mAudioMixer->setBufferProvider(name, track); 2736 mAudioMixer->enable(name); 2737 2738 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2739 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2740 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2741 mAudioMixer->setParameter( 2742 name, 2743 AudioMixer::TRACK, 2744 AudioMixer::FORMAT, (void *)track->format()); 2745 mAudioMixer->setParameter( 2746 name, 2747 AudioMixer::TRACK, 2748 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2749 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 2750 uint32_t maxSampleRate = mSampleRate * 2; 2751 uint32_t reqSampleRate = track->mServerProxy->getSampleRate(); 2752 if (reqSampleRate == 0) { 2753 reqSampleRate = mSampleRate; 2754 } else if (reqSampleRate > maxSampleRate) { 2755 reqSampleRate = maxSampleRate; 2756 } 2757 mAudioMixer->setParameter( 2758 name, 2759 AudioMixer::RESAMPLE, 2760 AudioMixer::SAMPLE_RATE, 2761 (void *)reqSampleRate); 2762 mAudioMixer->setParameter( 2763 name, 2764 AudioMixer::TRACK, 2765 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2766 mAudioMixer->setParameter( 2767 name, 2768 AudioMixer::TRACK, 2769 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2770 2771 // reset retry count 2772 track->mRetryCount = kMaxTrackRetries; 2773 2774 // If one track is ready, set the mixer ready if: 2775 // - the mixer was not ready during previous round OR 2776 // - no other track is not ready 2777 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2778 mixerStatus != MIXER_TRACKS_ENABLED) { 2779 mixerStatus = MIXER_TRACKS_READY; 2780 } 2781 } else { 2782 // clear effect chain input buffer if an active track underruns to avoid sending 2783 // previous audio buffer again to effects 2784 chain = getEffectChain_l(track->sessionId()); 2785 if (chain != 0) { 2786 chain->clearInputBuffer(); 2787 } 2788 2789 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2790 cblk->server, this); 2791 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2792 track->isStopped() || track->isPaused()) { 2793 // We have consumed all the buffers of this track. 2794 // Remove it from the list of active tracks. 2795 // TODO: use actual buffer filling status instead of latency when available from 2796 // audio HAL 2797 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2798 size_t framesWritten = mBytesWritten / mFrameSize; 2799 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2800 if (track->isStopped()) { 2801 track->reset(); 2802 } 2803 tracksToRemove->add(track); 2804 } 2805 } else { 2806 track->mUnderrunCount++; 2807 // No buffers for this track. Give it a few chances to 2808 // fill a buffer, then remove it from active list. 2809 if (--(track->mRetryCount) <= 0) { 2810 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2811 tracksToRemove->add(track); 2812 // indicate to client process that the track was disabled because of underrun; 2813 // it will then automatically call start() when data is available 2814 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2815 // If one track is not ready, mark the mixer also not ready if: 2816 // - the mixer was ready during previous round OR 2817 // - no other track is ready 2818 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2819 mixerStatus != MIXER_TRACKS_READY) { 2820 mixerStatus = MIXER_TRACKS_ENABLED; 2821 } 2822 } 2823 mAudioMixer->disable(name); 2824 } 2825 2826 } // local variable scope to avoid goto warning 2827track_is_ready: ; 2828 2829 } 2830 2831 // Push the new FastMixer state if necessary 2832 bool pauseAudioWatchdog = false; 2833 if (didModify) { 2834 state->mFastTracksGen++; 2835 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2836 if (kUseFastMixer == FastMixer_Dynamic && 2837 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2838 state->mCommand = FastMixerState::COLD_IDLE; 2839 state->mColdFutexAddr = &mFastMixerFutex; 2840 state->mColdGen++; 2841 mFastMixerFutex = 0; 2842 if (kUseFastMixer == FastMixer_Dynamic) { 2843 mNormalSink = mOutputSink; 2844 } 2845 // If we go into cold idle, need to wait for acknowledgement 2846 // so that fast mixer stops doing I/O. 2847 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2848 pauseAudioWatchdog = true; 2849 } 2850 } 2851 if (sq != NULL) { 2852 unsigned trackMask = state->mTrackMask; 2853 sq->end(didModify); 2854 if (didModify) { 2855 mNBLogWriter->logTimestamp(); 2856 mNBLogWriter->logf("push trackMask=%#x block=%d", trackMask, block); 2857 } 2858 sq->push(block); 2859 if (didModify) { 2860 mNBLogWriter->logTimestamp(); 2861 mNBLogWriter->log("pushed"); 2862 } 2863 } 2864#ifdef AUDIO_WATCHDOG 2865 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2866 mAudioWatchdog->pause(); 2867 } 2868#endif 2869 2870 // Now perform the deferred reset on fast tracks that have stopped 2871 while (resetMask != 0) { 2872 size_t i = __builtin_ctz(resetMask); 2873 ALOG_ASSERT(i < count); 2874 resetMask &= ~(1 << i); 2875 sp<Track> t = mActiveTracks[i].promote(); 2876 if (t == 0) { 2877 continue; 2878 } 2879 Track* track = t.get(); 2880 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2881 track->reset(); 2882 } 2883 2884 // remove all the tracks that need to be... 2885 count = tracksToRemove->size(); 2886 if (CC_UNLIKELY(count)) { 2887 for (size_t i=0 ; i<count ; i++) { 2888 const sp<Track>& track = tracksToRemove->itemAt(i); 2889 mNBLogWriter->logTimestamp(); 2890 mNBLogWriter->logf("prepareTracks_l remove name=%u mFastIndex=%d", track->name(), 2891 track->mFastIndex); 2892 mActiveTracks.remove(track); 2893 if (track->mainBuffer() != mMixBuffer) { 2894 chain = getEffectChain_l(track->sessionId()); 2895 if (chain != 0) { 2896 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2897 track->sessionId()); 2898 chain->decActiveTrackCnt(); 2899 } 2900 } 2901 if (track->isTerminated()) { 2902 removeTrack_l(track); 2903 } 2904 } 2905 } 2906 2907 // mix buffer must be cleared if all tracks are connected to an 2908 // effect chain as in this case the mixer will not write to 2909 // mix buffer and track effects will accumulate into it 2910 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2911 (mixedTracks == 0 && fastTracks > 0)) { 2912 // FIXME as a performance optimization, should remember previous zero status 2913 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2914 } 2915 2916 // if any fast tracks, then status is ready 2917 mMixerStatusIgnoringFastTracks = mixerStatus; 2918 if (fastTracks > 0) { 2919 mixerStatus = MIXER_TRACKS_READY; 2920 } 2921 return mixerStatus; 2922} 2923 2924// getTrackName_l() must be called with ThreadBase::mLock held 2925int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2926{ 2927 return mAudioMixer->getTrackName(channelMask, sessionId); 2928} 2929 2930// deleteTrackName_l() must be called with ThreadBase::mLock held 2931void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2932{ 2933 ALOGV("remove track (%d) and delete from mixer", name); 2934 mAudioMixer->deleteTrackName(name); 2935} 2936 2937// checkForNewParameters_l() must be called with ThreadBase::mLock held 2938bool AudioFlinger::MixerThread::checkForNewParameters_l() 2939{ 2940 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2941 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2942 bool reconfig = false; 2943 2944 while (!mNewParameters.isEmpty()) { 2945 2946 if (mFastMixer != NULL) { 2947 FastMixerStateQueue *sq = mFastMixer->sq(); 2948 FastMixerState *state = sq->begin(); 2949 if (!(state->mCommand & FastMixerState::IDLE)) { 2950 previousCommand = state->mCommand; 2951 state->mCommand = FastMixerState::HOT_IDLE; 2952 sq->end(); 2953 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2954 } else { 2955 sq->end(false /*didModify*/); 2956 } 2957 } 2958 2959 status_t status = NO_ERROR; 2960 String8 keyValuePair = mNewParameters[0]; 2961 AudioParameter param = AudioParameter(keyValuePair); 2962 int value; 2963 2964 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2965 reconfig = true; 2966 } 2967 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2968 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2969 status = BAD_VALUE; 2970 } else { 2971 reconfig = true; 2972 } 2973 } 2974 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2975 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2976 status = BAD_VALUE; 2977 } else { 2978 reconfig = true; 2979 } 2980 } 2981 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2982 // do not accept frame count changes if tracks are open as the track buffer 2983 // size depends on frame count and correct behavior would not be guaranteed 2984 // if frame count is changed after track creation 2985 if (!mTracks.isEmpty()) { 2986 status = INVALID_OPERATION; 2987 } else { 2988 reconfig = true; 2989 } 2990 } 2991 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2992#ifdef ADD_BATTERY_DATA 2993 // when changing the audio output device, call addBatteryData to notify 2994 // the change 2995 if (mOutDevice != value) { 2996 uint32_t params = 0; 2997 // check whether speaker is on 2998 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2999 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3000 } 3001 3002 audio_devices_t deviceWithoutSpeaker 3003 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3004 // check if any other device (except speaker) is on 3005 if (value & deviceWithoutSpeaker ) { 3006 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3007 } 3008 3009 if (params != 0) { 3010 addBatteryData(params); 3011 } 3012 } 3013#endif 3014 3015 // forward device change to effects that have requested to be 3016 // aware of attached audio device. 3017 mOutDevice = value; 3018 for (size_t i = 0; i < mEffectChains.size(); i++) { 3019 mEffectChains[i]->setDevice_l(mOutDevice); 3020 } 3021 } 3022 3023 if (status == NO_ERROR) { 3024 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3025 keyValuePair.string()); 3026 if (!mStandby && status == INVALID_OPERATION) { 3027 mOutput->stream->common.standby(&mOutput->stream->common); 3028 mStandby = true; 3029 mBytesWritten = 0; 3030 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3031 keyValuePair.string()); 3032 } 3033 if (status == NO_ERROR && reconfig) { 3034 delete mAudioMixer; 3035 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3036 mAudioMixer = NULL; 3037 readOutputParameters(); 3038 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3039 for (size_t i = 0; i < mTracks.size() ; i++) { 3040 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3041 if (name < 0) { 3042 break; 3043 } 3044 mTracks[i]->mName = name; 3045 } 3046 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3047 } 3048 } 3049 3050 mNewParameters.removeAt(0); 3051 3052 mParamStatus = status; 3053 mParamCond.signal(); 3054 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3055 // already timed out waiting for the status and will never signal the condition. 3056 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3057 } 3058 3059 if (!(previousCommand & FastMixerState::IDLE)) { 3060 ALOG_ASSERT(mFastMixer != NULL); 3061 FastMixerStateQueue *sq = mFastMixer->sq(); 3062 FastMixerState *state = sq->begin(); 3063 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3064 state->mCommand = previousCommand; 3065 sq->end(); 3066 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3067 } 3068 3069 return reconfig; 3070} 3071 3072 3073void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3074{ 3075 const size_t SIZE = 256; 3076 char buffer[SIZE]; 3077 String8 result; 3078 3079 PlaybackThread::dumpInternals(fd, args); 3080 3081 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3082 result.append(buffer); 3083 write(fd, result.string(), result.size()); 3084 3085 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3086 FastMixerDumpState copy = mFastMixerDumpState; 3087 copy.dump(fd); 3088 3089#ifdef STATE_QUEUE_DUMP 3090 // Similar for state queue 3091 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3092 observerCopy.dump(fd); 3093 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3094 mutatorCopy.dump(fd); 3095#endif 3096 3097 // Write the tee output to a .wav file 3098 dumpTee(fd, mTeeSource, mId); 3099 3100#ifdef AUDIO_WATCHDOG 3101 if (mAudioWatchdog != 0) { 3102 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3103 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3104 wdCopy.dump(fd); 3105 } 3106#endif 3107} 3108 3109uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3110{ 3111 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3112} 3113 3114uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3115{ 3116 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3117} 3118 3119void AudioFlinger::MixerThread::cacheParameters_l() 3120{ 3121 PlaybackThread::cacheParameters_l(); 3122 3123 // FIXME: Relaxed timing because of a certain device that can't meet latency 3124 // Should be reduced to 2x after the vendor fixes the driver issue 3125 // increase threshold again due to low power audio mode. The way this warning 3126 // threshold is calculated and its usefulness should be reconsidered anyway. 3127 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3128} 3129 3130// ---------------------------------------------------------------------------- 3131 3132AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3133 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3134 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3135 // mLeftVolFloat, mRightVolFloat 3136{ 3137} 3138 3139AudioFlinger::DirectOutputThread::~DirectOutputThread() 3140{ 3141} 3142 3143AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3144 Vector< sp<Track> > *tracksToRemove 3145) 3146{ 3147 sp<Track> trackToRemove; 3148 3149 mixer_state mixerStatus = MIXER_IDLE; 3150 3151 // find out which tracks need to be processed 3152 if (mActiveTracks.size() != 0) { 3153 sp<Track> t = mActiveTracks[0].promote(); 3154 // The track died recently 3155 if (t == 0) { 3156 return MIXER_IDLE; 3157 } 3158 3159 Track* const track = t.get(); 3160 audio_track_cblk_t* cblk = track->cblk(); 3161 3162 // The first time a track is added we wait 3163 // for all its buffers to be filled before processing it 3164 uint32_t minFrames; 3165 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3166 minFrames = mNormalFrameCount; 3167 } else { 3168 minFrames = 1; 3169 } 3170 if ((track->framesReady() >= minFrames) && track->isReady() && 3171 !track->isPaused() && !track->isTerminated()) 3172 { 3173 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3174 3175 if (track->mFillingUpStatus == Track::FS_FILLED) { 3176 track->mFillingUpStatus = Track::FS_ACTIVE; 3177 mLeftVolFloat = mRightVolFloat = 0; 3178 if (track->mState == TrackBase::RESUMING) { 3179 track->mState = TrackBase::ACTIVE; 3180 } 3181 } 3182 3183 // compute volume for this track 3184 float left, right; 3185 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { 3186 left = right = 0; 3187 if (track->isPausing()) { 3188 track->setPaused(); 3189 } 3190 } else { 3191 float typeVolume = mStreamTypes[track->streamType()].volume; 3192 float v = mMasterVolume * typeVolume; 3193 uint32_t vlr = track->mServerProxy->getVolumeLR(); 3194 float v_clamped = v * (vlr & 0xFFFF); 3195 if (v_clamped > MAX_GAIN) { 3196 v_clamped = MAX_GAIN; 3197 } 3198 left = v_clamped/MAX_GAIN; 3199 v_clamped = v * (vlr >> 16); 3200 if (v_clamped > MAX_GAIN) { 3201 v_clamped = MAX_GAIN; 3202 } 3203 right = v_clamped/MAX_GAIN; 3204 } 3205 3206 if (left != mLeftVolFloat || right != mRightVolFloat) { 3207 mLeftVolFloat = left; 3208 mRightVolFloat = right; 3209 3210 // Convert volumes from float to 8.24 3211 uint32_t vl = (uint32_t)(left * (1 << 24)); 3212 uint32_t vr = (uint32_t)(right * (1 << 24)); 3213 3214 // Delegate volume control to effect in track effect chain if needed 3215 // only one effect chain can be present on DirectOutputThread, so if 3216 // there is one, the track is connected to it 3217 if (!mEffectChains.isEmpty()) { 3218 // Do not ramp volume if volume is controlled by effect 3219 mEffectChains[0]->setVolume_l(&vl, &vr); 3220 left = (float)vl / (1 << 24); 3221 right = (float)vr / (1 << 24); 3222 } 3223 mOutput->stream->set_volume(mOutput->stream, left, right); 3224 } 3225 3226 // reset retry count 3227 track->mRetryCount = kMaxTrackRetriesDirect; 3228 mActiveTrack = t; 3229 mixerStatus = MIXER_TRACKS_READY; 3230 } else { 3231 // clear effect chain input buffer if an active track underruns to avoid sending 3232 // previous audio buffer again to effects 3233 if (!mEffectChains.isEmpty()) { 3234 mEffectChains[0]->clearInputBuffer(); 3235 } 3236 3237 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3238 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3239 track->isStopped() || track->isPaused()) { 3240 // We have consumed all the buffers of this track. 3241 // Remove it from the list of active tracks. 3242 // TODO: implement behavior for compressed audio 3243 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3244 size_t framesWritten = mBytesWritten / mFrameSize; 3245 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3246 if (track->isStopped()) { 3247 track->reset(); 3248 } 3249 trackToRemove = track; 3250 } 3251 } else { 3252 // No buffers for this track. Give it a few chances to 3253 // fill a buffer, then remove it from active list. 3254 if (--(track->mRetryCount) <= 0) { 3255 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3256 trackToRemove = track; 3257 } else { 3258 mixerStatus = MIXER_TRACKS_ENABLED; 3259 } 3260 } 3261 } 3262 } 3263 3264 // FIXME merge this with similar code for removing multiple tracks 3265 // remove all the tracks that need to be... 3266 if (CC_UNLIKELY(trackToRemove != 0)) { 3267 tracksToRemove->add(trackToRemove); 3268#if 0 3269 mNBLogWriter->logf("prepareTracks_l remove name=%u", trackToRemove->name()); 3270#endif 3271 mActiveTracks.remove(trackToRemove); 3272 if (!mEffectChains.isEmpty()) { 3273 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3274 trackToRemove->sessionId()); 3275 mEffectChains[0]->decActiveTrackCnt(); 3276 } 3277 if (trackToRemove->isTerminated()) { 3278 removeTrack_l(trackToRemove); 3279 } 3280 } 3281 3282 return mixerStatus; 3283} 3284 3285void AudioFlinger::DirectOutputThread::threadLoop_mix() 3286{ 3287 AudioBufferProvider::Buffer buffer; 3288 size_t frameCount = mFrameCount; 3289 int8_t *curBuf = (int8_t *)mMixBuffer; 3290 // output audio to hardware 3291 while (frameCount) { 3292 buffer.frameCount = frameCount; 3293 mActiveTrack->getNextBuffer(&buffer); 3294 if (CC_UNLIKELY(buffer.raw == NULL)) { 3295 memset(curBuf, 0, frameCount * mFrameSize); 3296 break; 3297 } 3298 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3299 frameCount -= buffer.frameCount; 3300 curBuf += buffer.frameCount * mFrameSize; 3301 mActiveTrack->releaseBuffer(&buffer); 3302 } 3303 sleepTime = 0; 3304 standbyTime = systemTime() + standbyDelay; 3305 mActiveTrack.clear(); 3306 3307} 3308 3309void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3310{ 3311 if (sleepTime == 0) { 3312 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3313 sleepTime = activeSleepTime; 3314 } else { 3315 sleepTime = idleSleepTime; 3316 } 3317 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3318 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3319 sleepTime = 0; 3320 } 3321} 3322 3323// getTrackName_l() must be called with ThreadBase::mLock held 3324int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3325 int sessionId) 3326{ 3327 return 0; 3328} 3329 3330// deleteTrackName_l() must be called with ThreadBase::mLock held 3331void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3332{ 3333} 3334 3335// checkForNewParameters_l() must be called with ThreadBase::mLock held 3336bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3337{ 3338 bool reconfig = false; 3339 3340 while (!mNewParameters.isEmpty()) { 3341 status_t status = NO_ERROR; 3342 String8 keyValuePair = mNewParameters[0]; 3343 AudioParameter param = AudioParameter(keyValuePair); 3344 int value; 3345 3346 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3347 // do not accept frame count changes if tracks are open as the track buffer 3348 // size depends on frame count and correct behavior would not be garantied 3349 // if frame count is changed after track creation 3350 if (!mTracks.isEmpty()) { 3351 status = INVALID_OPERATION; 3352 } else { 3353 reconfig = true; 3354 } 3355 } 3356 if (status == NO_ERROR) { 3357 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3358 keyValuePair.string()); 3359 if (!mStandby && status == INVALID_OPERATION) { 3360 mOutput->stream->common.standby(&mOutput->stream->common); 3361 mStandby = true; 3362 mBytesWritten = 0; 3363 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3364 keyValuePair.string()); 3365 } 3366 if (status == NO_ERROR && reconfig) { 3367 readOutputParameters(); 3368 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3369 } 3370 } 3371 3372 mNewParameters.removeAt(0); 3373 3374 mParamStatus = status; 3375 mParamCond.signal(); 3376 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3377 // already timed out waiting for the status and will never signal the condition. 3378 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3379 } 3380 return reconfig; 3381} 3382 3383uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3384{ 3385 uint32_t time; 3386 if (audio_is_linear_pcm(mFormat)) { 3387 time = PlaybackThread::activeSleepTimeUs(); 3388 } else { 3389 time = 10000; 3390 } 3391 return time; 3392} 3393 3394uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3395{ 3396 uint32_t time; 3397 if (audio_is_linear_pcm(mFormat)) { 3398 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3399 } else { 3400 time = 10000; 3401 } 3402 return time; 3403} 3404 3405uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3406{ 3407 uint32_t time; 3408 if (audio_is_linear_pcm(mFormat)) { 3409 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3410 } else { 3411 time = 10000; 3412 } 3413 return time; 3414} 3415 3416void AudioFlinger::DirectOutputThread::cacheParameters_l() 3417{ 3418 PlaybackThread::cacheParameters_l(); 3419 3420 // use shorter standby delay as on normal output to release 3421 // hardware resources as soon as possible 3422 standbyDelay = microseconds(activeSleepTime*2); 3423} 3424 3425// ---------------------------------------------------------------------------- 3426 3427AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3428 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3429 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3430 DUPLICATING), 3431 mWaitTimeMs(UINT_MAX) 3432{ 3433 addOutputTrack(mainThread); 3434} 3435 3436AudioFlinger::DuplicatingThread::~DuplicatingThread() 3437{ 3438 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3439 mOutputTracks[i]->destroy(); 3440 } 3441} 3442 3443void AudioFlinger::DuplicatingThread::threadLoop_mix() 3444{ 3445 // mix buffers... 3446 if (outputsReady(outputTracks)) { 3447 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3448 } else { 3449 memset(mMixBuffer, 0, mixBufferSize); 3450 } 3451 sleepTime = 0; 3452 writeFrames = mNormalFrameCount; 3453 standbyTime = systemTime() + standbyDelay; 3454} 3455 3456void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3457{ 3458 if (sleepTime == 0) { 3459 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3460 sleepTime = activeSleepTime; 3461 } else { 3462 sleepTime = idleSleepTime; 3463 } 3464 } else if (mBytesWritten != 0) { 3465 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3466 writeFrames = mNormalFrameCount; 3467 memset(mMixBuffer, 0, mixBufferSize); 3468 } else { 3469 // flush remaining overflow buffers in output tracks 3470 writeFrames = 0; 3471 } 3472 sleepTime = 0; 3473 } 3474} 3475 3476void AudioFlinger::DuplicatingThread::threadLoop_write() 3477{ 3478 for (size_t i = 0; i < outputTracks.size(); i++) { 3479 outputTracks[i]->write(mMixBuffer, writeFrames); 3480 } 3481 mBytesWritten += mixBufferSize; 3482} 3483 3484void AudioFlinger::DuplicatingThread::threadLoop_standby() 3485{ 3486 // DuplicatingThread implements standby by stopping all tracks 3487 for (size_t i = 0; i < outputTracks.size(); i++) { 3488 outputTracks[i]->stop(); 3489 } 3490} 3491 3492void AudioFlinger::DuplicatingThread::saveOutputTracks() 3493{ 3494 outputTracks = mOutputTracks; 3495} 3496 3497void AudioFlinger::DuplicatingThread::clearOutputTracks() 3498{ 3499 outputTracks.clear(); 3500} 3501 3502void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3503{ 3504 Mutex::Autolock _l(mLock); 3505 // FIXME explain this formula 3506 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3507 OutputTrack *outputTrack = new OutputTrack(thread, 3508 this, 3509 mSampleRate, 3510 mFormat, 3511 mChannelMask, 3512 frameCount); 3513 if (outputTrack->cblk() != NULL) { 3514 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3515 mOutputTracks.add(outputTrack); 3516 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3517 updateWaitTime_l(); 3518 } 3519} 3520 3521void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3522{ 3523 Mutex::Autolock _l(mLock); 3524 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3525 if (mOutputTracks[i]->thread() == thread) { 3526 mOutputTracks[i]->destroy(); 3527 mOutputTracks.removeAt(i); 3528 updateWaitTime_l(); 3529 return; 3530 } 3531 } 3532 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3533} 3534 3535// caller must hold mLock 3536void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3537{ 3538 mWaitTimeMs = UINT_MAX; 3539 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3540 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3541 if (strong != 0) { 3542 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3543 if (waitTimeMs < mWaitTimeMs) { 3544 mWaitTimeMs = waitTimeMs; 3545 } 3546 } 3547 } 3548} 3549 3550 3551bool AudioFlinger::DuplicatingThread::outputsReady( 3552 const SortedVector< sp<OutputTrack> > &outputTracks) 3553{ 3554 for (size_t i = 0; i < outputTracks.size(); i++) { 3555 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3556 if (thread == 0) { 3557 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3558 outputTracks[i].get()); 3559 return false; 3560 } 3561 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3562 // see note at standby() declaration 3563 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3564 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3565 thread.get()); 3566 return false; 3567 } 3568 } 3569 return true; 3570} 3571 3572uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3573{ 3574 return (mWaitTimeMs * 1000) / 2; 3575} 3576 3577void AudioFlinger::DuplicatingThread::cacheParameters_l() 3578{ 3579 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3580 updateWaitTime_l(); 3581 3582 MixerThread::cacheParameters_l(); 3583} 3584 3585// ---------------------------------------------------------------------------- 3586// Record 3587// ---------------------------------------------------------------------------- 3588 3589AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3590 AudioStreamIn *input, 3591 uint32_t sampleRate, 3592 audio_channel_mask_t channelMask, 3593 audio_io_handle_t id, 3594 audio_devices_t outDevice, 3595 audio_devices_t inDevice, 3596 const sp<NBAIO_Sink>& teeSink) : 3597 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 3598 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3599 // mRsmpInIndex and mInputBytes set by readInputParameters() 3600 mReqChannelCount(popcount(channelMask)), 3601 mReqSampleRate(sampleRate), 3602 // mBytesRead is only meaningful while active, and so is cleared in start() 3603 // (but might be better to also clear here for dump?) 3604 mTeeSink(teeSink) 3605{ 3606 snprintf(mName, kNameLength, "AudioIn_%X", id); 3607 3608 readInputParameters(); 3609 3610} 3611 3612 3613AudioFlinger::RecordThread::~RecordThread() 3614{ 3615 delete[] mRsmpInBuffer; 3616 delete mResampler; 3617 delete[] mRsmpOutBuffer; 3618} 3619 3620void AudioFlinger::RecordThread::onFirstRef() 3621{ 3622 run(mName, PRIORITY_URGENT_AUDIO); 3623} 3624 3625status_t AudioFlinger::RecordThread::readyToRun() 3626{ 3627 status_t status = initCheck(); 3628 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3629 return status; 3630} 3631 3632bool AudioFlinger::RecordThread::threadLoop() 3633{ 3634 AudioBufferProvider::Buffer buffer; 3635 sp<RecordTrack> activeTrack; 3636 Vector< sp<EffectChain> > effectChains; 3637 3638 nsecs_t lastWarning = 0; 3639 3640 inputStandBy(); 3641 acquireWakeLock(); 3642 3643 // used to verify we've read at least once before evaluating how many bytes were read 3644 bool readOnce = false; 3645 3646 // start recording 3647 while (!exitPending()) { 3648 3649 processConfigEvents(); 3650 3651 { // scope for mLock 3652 Mutex::Autolock _l(mLock); 3653 checkForNewParameters_l(); 3654 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3655 standby(); 3656 3657 if (exitPending()) { 3658 break; 3659 } 3660 3661 releaseWakeLock_l(); 3662 ALOGV("RecordThread: loop stopping"); 3663 // go to sleep 3664 mWaitWorkCV.wait(mLock); 3665 ALOGV("RecordThread: loop starting"); 3666 acquireWakeLock_l(); 3667 continue; 3668 } 3669 if (mActiveTrack != 0) { 3670 if (mActiveTrack->mState == TrackBase::PAUSING) { 3671 standby(); 3672 mActiveTrack.clear(); 3673 mStartStopCond.broadcast(); 3674 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3675 if (mReqChannelCount != mActiveTrack->channelCount()) { 3676 mActiveTrack.clear(); 3677 mStartStopCond.broadcast(); 3678 } else if (readOnce) { 3679 // record start succeeds only if first read from audio input 3680 // succeeds 3681 if (mBytesRead >= 0) { 3682 mActiveTrack->mState = TrackBase::ACTIVE; 3683 } else { 3684 mActiveTrack.clear(); 3685 } 3686 mStartStopCond.broadcast(); 3687 } 3688 mStandby = false; 3689 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3690 removeTrack_l(mActiveTrack); 3691 mActiveTrack.clear(); 3692 } 3693 } 3694 lockEffectChains_l(effectChains); 3695 } 3696 3697 if (mActiveTrack != 0) { 3698 if (mActiveTrack->mState != TrackBase::ACTIVE && 3699 mActiveTrack->mState != TrackBase::RESUMING) { 3700 unlockEffectChains(effectChains); 3701 usleep(kRecordThreadSleepUs); 3702 continue; 3703 } 3704 for (size_t i = 0; i < effectChains.size(); i ++) { 3705 effectChains[i]->process_l(); 3706 } 3707 3708 buffer.frameCount = mFrameCount; 3709 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3710 readOnce = true; 3711 size_t framesOut = buffer.frameCount; 3712 if (mResampler == NULL) { 3713 // no resampling 3714 while (framesOut) { 3715 size_t framesIn = mFrameCount - mRsmpInIndex; 3716 if (framesIn) { 3717 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3718 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3719 mActiveTrack->mFrameSize; 3720 if (framesIn > framesOut) 3721 framesIn = framesOut; 3722 mRsmpInIndex += framesIn; 3723 framesOut -= framesIn; 3724 if (mChannelCount == mReqChannelCount || 3725 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3726 memcpy(dst, src, framesIn * mFrameSize); 3727 } else { 3728 if (mChannelCount == 1) { 3729 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3730 (int16_t *)src, framesIn); 3731 } else { 3732 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3733 (int16_t *)src, framesIn); 3734 } 3735 } 3736 } 3737 if (framesOut && mFrameCount == mRsmpInIndex) { 3738 void *readInto; 3739 if (framesOut == mFrameCount && 3740 (mChannelCount == mReqChannelCount || 3741 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3742 readInto = buffer.raw; 3743 framesOut = 0; 3744 } else { 3745 readInto = mRsmpInBuffer; 3746 mRsmpInIndex = 0; 3747 } 3748 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 3749 if (mBytesRead <= 0) { 3750 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3751 { 3752 ALOGE("Error reading audio input"); 3753 // Force input into standby so that it tries to 3754 // recover at next read attempt 3755 inputStandBy(); 3756 usleep(kRecordThreadSleepUs); 3757 } 3758 mRsmpInIndex = mFrameCount; 3759 framesOut = 0; 3760 buffer.frameCount = 0; 3761 } else if (mTeeSink != 0) { 3762 (void) mTeeSink->write(readInto, 3763 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3764 } 3765 } 3766 } 3767 } else { 3768 // resampling 3769 3770 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3771 // alter output frame count as if we were expecting stereo samples 3772 if (mChannelCount == 1 && mReqChannelCount == 1) { 3773 framesOut >>= 1; 3774 } 3775 mResampler->resample(mRsmpOutBuffer, framesOut, 3776 this /* AudioBufferProvider* */); 3777 // ditherAndClamp() works as long as all buffers returned by 3778 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3779 if (mChannelCount == 2 && mReqChannelCount == 1) { 3780 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3781 // the resampler always outputs stereo samples: 3782 // do post stereo to mono conversion 3783 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3784 framesOut); 3785 } else { 3786 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3787 } 3788 3789 } 3790 if (mFramestoDrop == 0) { 3791 mActiveTrack->releaseBuffer(&buffer); 3792 } else { 3793 if (mFramestoDrop > 0) { 3794 mFramestoDrop -= buffer.frameCount; 3795 if (mFramestoDrop <= 0) { 3796 clearSyncStartEvent(); 3797 } 3798 } else { 3799 mFramestoDrop += buffer.frameCount; 3800 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3801 mSyncStartEvent->isCancelled()) { 3802 ALOGW("Synced record %s, session %d, trigger session %d", 3803 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3804 mActiveTrack->sessionId(), 3805 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3806 clearSyncStartEvent(); 3807 } 3808 } 3809 } 3810 mActiveTrack->clearOverflow(); 3811 } 3812 // client isn't retrieving buffers fast enough 3813 else { 3814 if (!mActiveTrack->setOverflow()) { 3815 nsecs_t now = systemTime(); 3816 if ((now - lastWarning) > kWarningThrottleNs) { 3817 ALOGW("RecordThread: buffer overflow"); 3818 lastWarning = now; 3819 } 3820 } 3821 // Release the processor for a while before asking for a new buffer. 3822 // This will give the application more chance to read from the buffer and 3823 // clear the overflow. 3824 usleep(kRecordThreadSleepUs); 3825 } 3826 } 3827 // enable changes in effect chain 3828 unlockEffectChains(effectChains); 3829 effectChains.clear(); 3830 } 3831 3832 standby(); 3833 3834 { 3835 Mutex::Autolock _l(mLock); 3836 mActiveTrack.clear(); 3837 mStartStopCond.broadcast(); 3838 } 3839 3840 releaseWakeLock(); 3841 3842 ALOGV("RecordThread %p exiting", this); 3843 return false; 3844} 3845 3846void AudioFlinger::RecordThread::standby() 3847{ 3848 if (!mStandby) { 3849 inputStandBy(); 3850 mStandby = true; 3851 } 3852} 3853 3854void AudioFlinger::RecordThread::inputStandBy() 3855{ 3856 mInput->stream->common.standby(&mInput->stream->common); 3857} 3858 3859sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3860 const sp<AudioFlinger::Client>& client, 3861 uint32_t sampleRate, 3862 audio_format_t format, 3863 audio_channel_mask_t channelMask, 3864 size_t frameCount, 3865 int sessionId, 3866 IAudioFlinger::track_flags_t flags, 3867 pid_t tid, 3868 status_t *status) 3869{ 3870 sp<RecordTrack> track; 3871 status_t lStatus; 3872 3873 lStatus = initCheck(); 3874 if (lStatus != NO_ERROR) { 3875 ALOGE("Audio driver not initialized."); 3876 goto Exit; 3877 } 3878 3879 // FIXME use flags and tid similar to createTrack_l() 3880 3881 { // scope for mLock 3882 Mutex::Autolock _l(mLock); 3883 3884 track = new RecordTrack(this, client, sampleRate, 3885 format, channelMask, frameCount, sessionId); 3886 3887 if (track->getCblk() == 0) { 3888 lStatus = NO_MEMORY; 3889 goto Exit; 3890 } 3891 mTracks.add(track); 3892 3893 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3894 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3895 mAudioFlinger->btNrecIsOff(); 3896 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3897 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3898 } 3899 lStatus = NO_ERROR; 3900 3901Exit: 3902 if (status) { 3903 *status = lStatus; 3904 } 3905 return track; 3906} 3907 3908status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3909 AudioSystem::sync_event_t event, 3910 int triggerSession) 3911{ 3912 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3913 sp<ThreadBase> strongMe = this; 3914 status_t status = NO_ERROR; 3915 3916 if (event == AudioSystem::SYNC_EVENT_NONE) { 3917 clearSyncStartEvent(); 3918 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3919 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3920 triggerSession, 3921 recordTrack->sessionId(), 3922 syncStartEventCallback, 3923 this); 3924 // Sync event can be cancelled by the trigger session if the track is not in a 3925 // compatible state in which case we start record immediately 3926 if (mSyncStartEvent->isCancelled()) { 3927 clearSyncStartEvent(); 3928 } else { 3929 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3930 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3931 } 3932 } 3933 3934 { 3935 AutoMutex lock(mLock); 3936 if (mActiveTrack != 0) { 3937 if (recordTrack != mActiveTrack.get()) { 3938 status = -EBUSY; 3939 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3940 mActiveTrack->mState = TrackBase::ACTIVE; 3941 } 3942 return status; 3943 } 3944 3945 recordTrack->mState = TrackBase::IDLE; 3946 mActiveTrack = recordTrack; 3947 mLock.unlock(); 3948 status_t status = AudioSystem::startInput(mId); 3949 mLock.lock(); 3950 if (status != NO_ERROR) { 3951 mActiveTrack.clear(); 3952 clearSyncStartEvent(); 3953 return status; 3954 } 3955 mRsmpInIndex = mFrameCount; 3956 mBytesRead = 0; 3957 if (mResampler != NULL) { 3958 mResampler->reset(); 3959 } 3960 mActiveTrack->mState = TrackBase::RESUMING; 3961 // signal thread to start 3962 ALOGV("Signal record thread"); 3963 mWaitWorkCV.broadcast(); 3964 // do not wait for mStartStopCond if exiting 3965 if (exitPending()) { 3966 mActiveTrack.clear(); 3967 status = INVALID_OPERATION; 3968 goto startError; 3969 } 3970 mStartStopCond.wait(mLock); 3971 if (mActiveTrack == 0) { 3972 ALOGV("Record failed to start"); 3973 status = BAD_VALUE; 3974 goto startError; 3975 } 3976 ALOGV("Record started OK"); 3977 return status; 3978 } 3979startError: 3980 AudioSystem::stopInput(mId); 3981 clearSyncStartEvent(); 3982 return status; 3983} 3984 3985void AudioFlinger::RecordThread::clearSyncStartEvent() 3986{ 3987 if (mSyncStartEvent != 0) { 3988 mSyncStartEvent->cancel(); 3989 } 3990 mSyncStartEvent.clear(); 3991 mFramestoDrop = 0; 3992} 3993 3994void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 3995{ 3996 sp<SyncEvent> strongEvent = event.promote(); 3997 3998 if (strongEvent != 0) { 3999 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4000 me->handleSyncStartEvent(strongEvent); 4001 } 4002} 4003 4004void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4005{ 4006 if (event == mSyncStartEvent) { 4007 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4008 // from audio HAL 4009 mFramestoDrop = mFrameCount * 2; 4010 } 4011} 4012 4013bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 4014 ALOGV("RecordThread::stop"); 4015 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4016 return false; 4017 } 4018 recordTrack->mState = TrackBase::PAUSING; 4019 // do not wait for mStartStopCond if exiting 4020 if (exitPending()) { 4021 return true; 4022 } 4023 mStartStopCond.wait(mLock); 4024 // if we have been restarted, recordTrack == mActiveTrack.get() here 4025 if (exitPending() || recordTrack != mActiveTrack.get()) { 4026 ALOGV("Record stopped OK"); 4027 return true; 4028 } 4029 return false; 4030} 4031 4032bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4033{ 4034 return false; 4035} 4036 4037status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4038{ 4039#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4040 if (!isValidSyncEvent(event)) { 4041 return BAD_VALUE; 4042 } 4043 4044 int eventSession = event->triggerSession(); 4045 status_t ret = NAME_NOT_FOUND; 4046 4047 Mutex::Autolock _l(mLock); 4048 4049 for (size_t i = 0; i < mTracks.size(); i++) { 4050 sp<RecordTrack> track = mTracks[i]; 4051 if (eventSession == track->sessionId()) { 4052 (void) track->setSyncEvent(event); 4053 ret = NO_ERROR; 4054 } 4055 } 4056 return ret; 4057#else 4058 return BAD_VALUE; 4059#endif 4060} 4061 4062// destroyTrack_l() must be called with ThreadBase::mLock held 4063void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4064{ 4065 track->mState = TrackBase::TERMINATED; 4066 // active tracks are removed by threadLoop() 4067 if (mActiveTrack != track) { 4068 removeTrack_l(track); 4069 } 4070} 4071 4072void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4073{ 4074 mTracks.remove(track); 4075 // need anything related to effects here? 4076} 4077 4078void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4079{ 4080 dumpInternals(fd, args); 4081 dumpTracks(fd, args); 4082 dumpEffectChains(fd, args); 4083} 4084 4085void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4086{ 4087 const size_t SIZE = 256; 4088 char buffer[SIZE]; 4089 String8 result; 4090 4091 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4092 result.append(buffer); 4093 4094 if (mActiveTrack != 0) { 4095 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4096 result.append(buffer); 4097 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4098 result.append(buffer); 4099 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4100 result.append(buffer); 4101 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4102 result.append(buffer); 4103 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4104 result.append(buffer); 4105 } else { 4106 result.append("No active record client\n"); 4107 } 4108 4109 write(fd, result.string(), result.size()); 4110 4111 dumpBase(fd, args); 4112} 4113 4114void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4115{ 4116 const size_t SIZE = 256; 4117 char buffer[SIZE]; 4118 String8 result; 4119 4120 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4121 result.append(buffer); 4122 RecordTrack::appendDumpHeader(result); 4123 for (size_t i = 0; i < mTracks.size(); ++i) { 4124 sp<RecordTrack> track = mTracks[i]; 4125 if (track != 0) { 4126 track->dump(buffer, SIZE); 4127 result.append(buffer); 4128 } 4129 } 4130 4131 if (mActiveTrack != 0) { 4132 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4133 result.append(buffer); 4134 RecordTrack::appendDumpHeader(result); 4135 mActiveTrack->dump(buffer, SIZE); 4136 result.append(buffer); 4137 4138 } 4139 write(fd, result.string(), result.size()); 4140} 4141 4142// AudioBufferProvider interface 4143status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4144{ 4145 size_t framesReq = buffer->frameCount; 4146 size_t framesReady = mFrameCount - mRsmpInIndex; 4147 int channelCount; 4148 4149 if (framesReady == 0) { 4150 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4151 if (mBytesRead <= 0) { 4152 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4153 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4154 // Force input into standby so that it tries to 4155 // recover at next read attempt 4156 inputStandBy(); 4157 usleep(kRecordThreadSleepUs); 4158 } 4159 buffer->raw = NULL; 4160 buffer->frameCount = 0; 4161 return NOT_ENOUGH_DATA; 4162 } 4163 mRsmpInIndex = 0; 4164 framesReady = mFrameCount; 4165 } 4166 4167 if (framesReq > framesReady) { 4168 framesReq = framesReady; 4169 } 4170 4171 if (mChannelCount == 1 && mReqChannelCount == 2) { 4172 channelCount = 1; 4173 } else { 4174 channelCount = 2; 4175 } 4176 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4177 buffer->frameCount = framesReq; 4178 return NO_ERROR; 4179} 4180 4181// AudioBufferProvider interface 4182void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4183{ 4184 mRsmpInIndex += buffer->frameCount; 4185 buffer->frameCount = 0; 4186} 4187 4188bool AudioFlinger::RecordThread::checkForNewParameters_l() 4189{ 4190 bool reconfig = false; 4191 4192 while (!mNewParameters.isEmpty()) { 4193 status_t status = NO_ERROR; 4194 String8 keyValuePair = mNewParameters[0]; 4195 AudioParameter param = AudioParameter(keyValuePair); 4196 int value; 4197 audio_format_t reqFormat = mFormat; 4198 uint32_t reqSamplingRate = mReqSampleRate; 4199 uint32_t reqChannelCount = mReqChannelCount; 4200 4201 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4202 reqSamplingRate = value; 4203 reconfig = true; 4204 } 4205 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4206 reqFormat = (audio_format_t) value; 4207 reconfig = true; 4208 } 4209 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4210 reqChannelCount = popcount(value); 4211 reconfig = true; 4212 } 4213 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4214 // do not accept frame count changes if tracks are open as the track buffer 4215 // size depends on frame count and correct behavior would not be guaranteed 4216 // if frame count is changed after track creation 4217 if (mActiveTrack != 0) { 4218 status = INVALID_OPERATION; 4219 } else { 4220 reconfig = true; 4221 } 4222 } 4223 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4224 // forward device change to effects that have requested to be 4225 // aware of attached audio device. 4226 for (size_t i = 0; i < mEffectChains.size(); i++) { 4227 mEffectChains[i]->setDevice_l(value); 4228 } 4229 4230 // store input device and output device but do not forward output device to audio HAL. 4231 // Note that status is ignored by the caller for output device 4232 // (see AudioFlinger::setParameters() 4233 if (audio_is_output_devices(value)) { 4234 mOutDevice = value; 4235 status = BAD_VALUE; 4236 } else { 4237 mInDevice = value; 4238 // disable AEC and NS if the device is a BT SCO headset supporting those 4239 // pre processings 4240 if (mTracks.size() > 0) { 4241 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4242 mAudioFlinger->btNrecIsOff(); 4243 for (size_t i = 0; i < mTracks.size(); i++) { 4244 sp<RecordTrack> track = mTracks[i]; 4245 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4246 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4247 } 4248 } 4249 } 4250 } 4251 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4252 mAudioSource != (audio_source_t)value) { 4253 // forward device change to effects that have requested to be 4254 // aware of attached audio device. 4255 for (size_t i = 0; i < mEffectChains.size(); i++) { 4256 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4257 } 4258 mAudioSource = (audio_source_t)value; 4259 } 4260 if (status == NO_ERROR) { 4261 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4262 keyValuePair.string()); 4263 if (status == INVALID_OPERATION) { 4264 inputStandBy(); 4265 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4266 keyValuePair.string()); 4267 } 4268 if (reconfig) { 4269 if (status == BAD_VALUE && 4270 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4271 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4272 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4273 <= (2 * reqSamplingRate)) && 4274 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4275 <= FCC_2 && 4276 (reqChannelCount <= FCC_2)) { 4277 status = NO_ERROR; 4278 } 4279 if (status == NO_ERROR) { 4280 readInputParameters(); 4281 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4282 } 4283 } 4284 } 4285 4286 mNewParameters.removeAt(0); 4287 4288 mParamStatus = status; 4289 mParamCond.signal(); 4290 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4291 // already timed out waiting for the status and will never signal the condition. 4292 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4293 } 4294 return reconfig; 4295} 4296 4297String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4298{ 4299 char *s; 4300 String8 out_s8 = String8(); 4301 4302 Mutex::Autolock _l(mLock); 4303 if (initCheck() != NO_ERROR) { 4304 return out_s8; 4305 } 4306 4307 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4308 out_s8 = String8(s); 4309 free(s); 4310 return out_s8; 4311} 4312 4313void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4314 AudioSystem::OutputDescriptor desc; 4315 void *param2 = NULL; 4316 4317 switch (event) { 4318 case AudioSystem::INPUT_OPENED: 4319 case AudioSystem::INPUT_CONFIG_CHANGED: 4320 desc.channels = mChannelMask; 4321 desc.samplingRate = mSampleRate; 4322 desc.format = mFormat; 4323 desc.frameCount = mFrameCount; 4324 desc.latency = 0; 4325 param2 = &desc; 4326 break; 4327 4328 case AudioSystem::INPUT_CLOSED: 4329 default: 4330 break; 4331 } 4332 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4333} 4334 4335void AudioFlinger::RecordThread::readInputParameters() 4336{ 4337 delete mRsmpInBuffer; 4338 // mRsmpInBuffer is always assigned a new[] below 4339 delete mRsmpOutBuffer; 4340 mRsmpOutBuffer = NULL; 4341 delete mResampler; 4342 mResampler = NULL; 4343 4344 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4345 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4346 mChannelCount = (uint16_t)popcount(mChannelMask); 4347 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4348 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4349 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4350 mFrameCount = mInputBytes / mFrameSize; 4351 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4352 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4353 4354 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4355 { 4356 int channelCount; 4357 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4358 // stereo to mono post process as the resampler always outputs stereo. 4359 if (mChannelCount == 1 && mReqChannelCount == 2) { 4360 channelCount = 1; 4361 } else { 4362 channelCount = 2; 4363 } 4364 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4365 mResampler->setSampleRate(mSampleRate); 4366 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4367 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4368 4369 // optmization: if mono to mono, alter input frame count as if we were inputing 4370 // stereo samples 4371 if (mChannelCount == 1 && mReqChannelCount == 1) { 4372 mFrameCount >>= 1; 4373 } 4374 4375 } 4376 mRsmpInIndex = mFrameCount; 4377} 4378 4379unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4380{ 4381 Mutex::Autolock _l(mLock); 4382 if (initCheck() != NO_ERROR) { 4383 return 0; 4384 } 4385 4386 return mInput->stream->get_input_frames_lost(mInput->stream); 4387} 4388 4389uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4390{ 4391 Mutex::Autolock _l(mLock); 4392 uint32_t result = 0; 4393 if (getEffectChain_l(sessionId) != 0) { 4394 result = EFFECT_SESSION; 4395 } 4396 4397 for (size_t i = 0; i < mTracks.size(); ++i) { 4398 if (sessionId == mTracks[i]->sessionId()) { 4399 result |= TRACK_SESSION; 4400 break; 4401 } 4402 } 4403 4404 return result; 4405} 4406 4407KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4408{ 4409 KeyedVector<int, bool> ids; 4410 Mutex::Autolock _l(mLock); 4411 for (size_t j = 0; j < mTracks.size(); ++j) { 4412 sp<RecordThread::RecordTrack> track = mTracks[j]; 4413 int sessionId = track->sessionId(); 4414 if (ids.indexOfKey(sessionId) < 0) { 4415 ids.add(sessionId, true); 4416 } 4417 } 4418 return ids; 4419} 4420 4421AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4422{ 4423 Mutex::Autolock _l(mLock); 4424 AudioStreamIn *input = mInput; 4425 mInput = NULL; 4426 return input; 4427} 4428 4429// this method must always be called either with ThreadBase mLock held or inside the thread loop 4430audio_stream_t* AudioFlinger::RecordThread::stream() const 4431{ 4432 if (mInput == NULL) { 4433 return NULL; 4434 } 4435 return &mInput->stream->common; 4436} 4437 4438status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4439{ 4440 // only one chain per input thread 4441 if (mEffectChains.size() != 0) { 4442 return INVALID_OPERATION; 4443 } 4444 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4445 4446 chain->setInBuffer(NULL); 4447 chain->setOutBuffer(NULL); 4448 4449 checkSuspendOnAddEffectChain_l(chain); 4450 4451 mEffectChains.add(chain); 4452 4453 return NO_ERROR; 4454} 4455 4456size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4457{ 4458 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4459 ALOGW_IF(mEffectChains.size() != 1, 4460 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4461 chain.get(), mEffectChains.size(), this); 4462 if (mEffectChains.size() == 1) { 4463 mEffectChains.removeAt(0); 4464 } 4465 return 0; 4466} 4467 4468}; // namespace android 4469