AudioFlinger.cpp revision a88ed026402d92d699c336aa11267616007e4a9d
1d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright/* 2d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright** 3d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright** Copyright 2007, The Android Open Source Project 4d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright** 5d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright** Licensed under the Apache License, Version 2.0 (the "License"); 6d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright** you may not use this file except in compliance with the License. 7d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright** You may obtain a copy of the License at 8d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright** 9d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright** http://www.apache.org/licenses/LICENSE-2.0 10d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright** 11d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright** Unless required by applicable law or agreed to in writing, software 12d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright** distributed under the License is distributed on an "AS IS" BASIS, 13d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright** See the License for the specific language governing permissions and 15d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright** limitations under the License. 16d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright*/ 17d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 18d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 19d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#define LOG_TAG "AudioFlinger" 20d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright//#define LOG_NDEBUG 0 21d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 22d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <math.h> 23d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <signal.h> 24d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <sys/time.h> 25d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <sys/resource.h> 26d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 27d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <binder/IPCThreadState.h> 28d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <binder/IServiceManager.h> 29d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <utils/Log.h> 30f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown#include <utils/Trace.h> 31f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown#include <binder/Parcel.h> 32f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown#include <binder/IPCThreadState.h> 33d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <utils/String16.h> 34d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <utils/threads.h> 35d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <utils/Atomic.h> 36d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 37d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <cutils/bitops.h> 38d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <cutils/properties.h> 39d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <cutils/compiler.h> 40d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 41d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#undef ADD_BATTERY_DATA 42d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 43d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#ifdef ADD_BATTERY_DATA 44d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <media/IMediaPlayerService.h> 45d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <media/IMediaDeathNotifier.h> 46d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#endif 47d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 48d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <private/media/AudioTrackShared.h> 49d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <private/media/AudioEffectShared.h> 50d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 51d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <system/audio.h> 5239efe3e5bf6282a4851e0eb3b938060c8f7790aeNarayan Kamath#include <hardware/audio.h> 53d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 54d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include "AudioMixer.h" 5539efe3e5bf6282a4851e0eb3b938060c8f7790aeNarayan Kamath#include "AudioFlinger.h" 5639efe3e5bf6282a4851e0eb3b938060c8f7790aeNarayan Kamath#include "ServiceUtilities.h" 5739efe3e5bf6282a4851e0eb3b938060c8f7790aeNarayan Kamath 58d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <media/EffectsFactoryApi.h> 59d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <audio_effects/effect_visualizer.h> 60d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <audio_effects/effect_ns.h> 6139efe3e5bf6282a4851e0eb3b938060c8f7790aeNarayan Kamath#include <audio_effects/effect_aec.h> 62d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 63d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <audio_utils/primitives.h> 64d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 65d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <powermanager/PowerManager.h> 66d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 67d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 6839efe3e5bf6282a4851e0eb3b938060c8f7790aeNarayan Kamath#ifdef DEBUG_CPU_USAGE 69d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <cpustats/CentralTendencyStatistics.h> 70d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <cpustats/ThreadCpuUsage.h> 71d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#endif 7239efe3e5bf6282a4851e0eb3b938060c8f7790aeNarayan Kamath 73d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <common_time/cc_helper.h> 74d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include <common_time/local_clock.h> 7539efe3e5bf6282a4851e0eb3b938060c8f7790aeNarayan Kamath 76d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include "FastMixer.h" 77d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 7839efe3e5bf6282a4851e0eb3b938060c8f7790aeNarayan Kamath// NBAIO implementations 7939efe3e5bf6282a4851e0eb3b938060c8f7790aeNarayan Kamath#include "AudioStreamOutSink.h" 80d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include "MonoPipe.h" 81d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include "MonoPipeReader.h" 82d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include "Pipe.h" 8339efe3e5bf6282a4851e0eb3b938060c8f7790aeNarayan Kamath#include "PipeReader.h" 8439efe3e5bf6282a4851e0eb3b938060c8f7790aeNarayan Kamath#include "SourceAudioBufferProvider.h" 85d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 86d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#ifdef HAVE_REQUEST_PRIORITY 87d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#include "SchedulingPolicyService.h" 8839efe3e5bf6282a4851e0eb3b938060c8f7790aeNarayan Kamath#endif 89d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 90d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#ifdef SOAKER 9139efe3e5bf6282a4851e0eb3b938060c8f7790aeNarayan Kamath#include "Soaker.h" 92d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#endif 93d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 9439efe3e5bf6282a4851e0eb3b938060c8f7790aeNarayan Kamath// ---------------------------------------------------------------------------- 95d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 96d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// Note: the following macro is used for extremely verbose logging message. In 97d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 98d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// 0; but one side effect of this is to turn all LOGV's as well. Some messages 99d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// are so verbose that we want to suppress them even when we have ALOG_ASSERT 100d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// turned on. Do not uncomment the #def below unless you really know what you 101d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// are doing and want to see all of the extremely verbose messages. 102d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright//#define VERY_VERY_VERBOSE_LOGGING 103d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#ifdef VERY_VERY_VERBOSE_LOGGING 104d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#define ALOGVV ALOGV 105d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#else 106d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#define ALOGVV(a...) do { } while(0) 107d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#endif 108d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 109d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightnamespace android { 110d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 111d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightstatic const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 112d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightstatic const char kHardwareLockedString[] = "Hardware lock is taken\n"; 113d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 114d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightstatic const float MAX_GAIN = 4096.0f; 115d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightstatic const uint32_t MAX_GAIN_INT = 0x1000; 116d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 117d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// retry counts for buffer fill timeout 118d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// 50 * ~20msecs = 1 second 119d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightstatic const int8_t kMaxTrackRetries = 50; 120d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightstatic const int8_t kMaxTrackStartupRetries = 50; 121d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// allow less retry attempts on direct output thread. 122d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// direct outputs can be a scarce resource in audio hardware and should 123d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// be released as quickly as possible. 124d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightstatic const int8_t kMaxTrackRetriesDirect = 2; 125d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 126f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brownstatic const int kDumpLockRetries = 50; 127f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brownstatic const int kDumpLockSleepUs = 20000; 128d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 129d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// don't warn about blocked writes or record buffer overflows more often than this 130d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightstatic const nsecs_t kWarningThrottleNs = seconds(5); 131d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 132d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// RecordThread loop sleep time upon application overrun or audio HAL read error 133d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightstatic const int kRecordThreadSleepUs = 5000; 134d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 135f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown// maximum time to wait for setParameters to complete 136f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brownstatic const nsecs_t kSetParametersTimeoutNs = seconds(2); 137d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 138d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// minimum sleep time for the mixer thread loop when tracks are active but in underrun 139d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightstatic const uint32_t kMinThreadSleepTimeUs = 5000; 140d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// maximum divider applied to the active sleep time in the mixer thread loop 141d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightstatic const uint32_t kMaxThreadSleepTimeShift = 2; 142d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 143d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// minimum normal mix buffer size, expressed in milliseconds rather than frames 144d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightstatic const uint32_t kMinNormalMixBufferSizeMs = 20; 145d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// maximum normal mix buffer size 146d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightstatic const uint32_t kMaxNormalMixBufferSizeMs = 24; 147d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 148d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightnsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 149d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 150d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// Whether to use fast mixer 151d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightstatic const enum { 152d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright FastMixer_Never, // never initialize or use: for debugging only 1537b159c9a4f589da7fdab7c16f3aefea25e0e7e4fMichael Wright FastMixer_Always, // always initialize and use, even if not needed: for debugging only 154d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright // normal mixer multiplier is 1 155d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright FastMixer_Static, // initialize if needed, then use all the time if initialized, 156f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown // multiplier is calculated based on min & max normal mixer buffer size 157f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 158d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright // multiplier is calculated based on min & max normal mixer buffer size 159d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright // FIXME for FastMixer_Dynamic: 160d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright // Supporting this option will require fixing HALs that can't handle large writes. 161d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright // For example, one HAL implementation returns an error from a large write, 162d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright // and another HAL implementation corrupts memory, possibly in the sample rate converter. 163d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright // We could either fix the HAL implementations, or provide a wrapper that breaks 1647b159c9a4f589da7fdab7c16f3aefea25e0e7e4fMichael Wright // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 165d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright} kUseFastMixer = FastMixer_Static; 166d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 167f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown// ---------------------------------------------------------------------------- 168f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown 169d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#ifdef ADD_BATTERY_DATA 170d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// To collect the amplifier usage 171d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightstatic void addBatteryData(uint32_t params) { 172d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 1738b10c65312c0cd9a76fe9bdae2917e13d1e120ebDan Albert if (service == NULL) { 1747b159c9a4f589da7fdab7c16f3aefea25e0e7e4fMichael Wright // it already logged 175d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright return; 176d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright } 177f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown 178f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown service->addBatteryData(params); 179d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright} 180d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright#endif 181d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 182d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightstatic int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 183d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright{ 184d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright const hw_module_t *mod; 1857b159c9a4f589da7fdab7c16f3aefea25e0e7e4fMichael Wright int rc; 186d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 187d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 188f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 189f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 190d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright if (rc) { 191d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright goto out; 192d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright } 193d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright rc = audio_hw_device_open(mod, dev); 1948b10c65312c0cd9a76fe9bdae2917e13d1e120ebDan Albert ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 1957b159c9a4f589da7fdab7c16f3aefea25e0e7e4fMichael Wright AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 196d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright if (rc) { 197d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright goto out; 198f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown } 199f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 200d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 201d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright rc = BAD_VALUE; 202d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright goto out; 203d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright } 204d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright return 0; 2057b159c9a4f589da7fdab7c16f3aefea25e0e7e4fMichael Wright 206d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightout: 207d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright *dev = NULL; 208f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown return rc; 209f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown} 210d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 211d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright// ---------------------------------------------------------------------------- 212d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 213d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael WrightAudioFlinger::AudioFlinger() 2147b159c9a4f589da7fdab7c16f3aefea25e0e7e4fMichael Wright : BnAudioFlinger(), 215d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright mPrimaryHardwareDev(NULL), 216d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 217f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown mMasterVolume(1.0f), 218f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown mMasterVolumeSupportLvl(MVS_NONE), 219d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright mMasterMute(false), 220d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright mNextUniqueId(1), 221d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright mMode(AUDIO_MODE_INVALID), 222d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright mBtNrecIsOff(false) 223d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright{ 224d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright} 2257b159c9a4f589da7fdab7c16f3aefea25e0e7e4fMichael Wright 226d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wrightvoid AudioFlinger::onFirstRef() 227d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright{ 228f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown int rc = 0; 229f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown 230d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright Mutex::Autolock _l(mLock); 231d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 232d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright /* TODO: move all this work into an Init() function */ 233d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright char val_str[PROPERTY_VALUE_MAX] = { 0 }; 234d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 2357b159c9a4f589da7fdab7c16f3aefea25e0e7e4fMichael Wright uint32_t int_val; 236d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright if (1 == sscanf(val_str, "%u", &int_val)) { 237d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright mStandbyTimeInNsecs = milliseconds(int_val); 238f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown ALOGI("Using %u mSec as standby time.", int_val); 239f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown } else { 240d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 241d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright ALOGI("Using default %u mSec as standby time.", 242d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright (uint32_t)(mStandbyTimeInNsecs / 1000000)); 243d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright } 244d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright } 245d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 246d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright mMode = AUDIO_MODE_NORMAL; 2477b159c9a4f589da7fdab7c16f3aefea25e0e7e4fMichael Wright mMasterVolumeSW = 1.0; 248d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright mMasterVolume = 1.0; 249d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright mHardwareStatus = AUDIO_HW_IDLE; 250f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown} 251f086ddbb97e59bd4a0c27745f6e6cc9832a2d4f8Jeff Brown 252d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael WrightAudioFlinger::~AudioFlinger() 253d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright{ 254d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright 255d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright while (!mRecordThreads.isEmpty()) { 256d02c5b6aace05d9fd938e2d03705ac4f60f8da19Michael Wright // closeInput() will remove first entry from mRecordThreads 257 closeInput(mRecordThreads.keyAt(0)); 258 } 259 while (!mPlaybackThreads.isEmpty()) { 260 // closeOutput() will remove first entry from mPlaybackThreads 261 closeOutput(mPlaybackThreads.keyAt(0)); 262 } 263 264 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 265 // no mHardwareLock needed, as there are no other references to this 266 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 267 delete mAudioHwDevs.valueAt(i); 268 } 269} 270 271static const char * const audio_interfaces[] = { 272 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 273 AUDIO_HARDWARE_MODULE_ID_A2DP, 274 AUDIO_HARDWARE_MODULE_ID_USB, 275}; 276#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 277 278audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 279{ 280 // if module is 0, the request comes from an old policy manager and we should load 281 // well known modules 282 if (module == 0) { 283 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 284 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 285 loadHwModule_l(audio_interfaces[i]); 286 } 287 } else { 288 // check a match for the requested module handle 289 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 290 if (audioHwdevice != NULL) { 291 return audioHwdevice->hwDevice(); 292 } 293 } 294 // then try to find a module supporting the requested device. 295 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 296 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 297 if ((dev->get_supported_devices(dev) & devices) == devices) 298 return dev; 299 } 300 301 return NULL; 302} 303 304status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 305{ 306 const size_t SIZE = 256; 307 char buffer[SIZE]; 308 String8 result; 309 310 result.append("Clients:\n"); 311 for (size_t i = 0; i < mClients.size(); ++i) { 312 sp<Client> client = mClients.valueAt(i).promote(); 313 if (client != 0) { 314 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 315 result.append(buffer); 316 } 317 } 318 319 result.append("Global session refs:\n"); 320 result.append(" session pid count\n"); 321 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 322 AudioSessionRef *r = mAudioSessionRefs[i]; 323 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 324 result.append(buffer); 325 } 326 write(fd, result.string(), result.size()); 327 return NO_ERROR; 328} 329 330 331status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 332{ 333 const size_t SIZE = 256; 334 char buffer[SIZE]; 335 String8 result; 336 hardware_call_state hardwareStatus = mHardwareStatus; 337 338 snprintf(buffer, SIZE, "Hardware status: %d\n" 339 "Standby Time mSec: %u\n", 340 hardwareStatus, 341 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 342 result.append(buffer); 343 write(fd, result.string(), result.size()); 344 return NO_ERROR; 345} 346 347status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 348{ 349 const size_t SIZE = 256; 350 char buffer[SIZE]; 351 String8 result; 352 snprintf(buffer, SIZE, "Permission Denial: " 353 "can't dump AudioFlinger from pid=%d, uid=%d\n", 354 IPCThreadState::self()->getCallingPid(), 355 IPCThreadState::self()->getCallingUid()); 356 result.append(buffer); 357 write(fd, result.string(), result.size()); 358 return NO_ERROR; 359} 360 361static bool tryLock(Mutex& mutex) 362{ 363 bool locked = false; 364 for (int i = 0; i < kDumpLockRetries; ++i) { 365 if (mutex.tryLock() == NO_ERROR) { 366 locked = true; 367 break; 368 } 369 usleep(kDumpLockSleepUs); 370 } 371 return locked; 372} 373 374status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 375{ 376 if (!dumpAllowed()) { 377 dumpPermissionDenial(fd, args); 378 } else { 379 // get state of hardware lock 380 bool hardwareLocked = tryLock(mHardwareLock); 381 if (!hardwareLocked) { 382 String8 result(kHardwareLockedString); 383 write(fd, result.string(), result.size()); 384 } else { 385 mHardwareLock.unlock(); 386 } 387 388 bool locked = tryLock(mLock); 389 390 // failed to lock - AudioFlinger is probably deadlocked 391 if (!locked) { 392 String8 result(kDeadlockedString); 393 write(fd, result.string(), result.size()); 394 } 395 396 dumpClients(fd, args); 397 dumpInternals(fd, args); 398 399 // dump playback threads 400 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 401 mPlaybackThreads.valueAt(i)->dump(fd, args); 402 } 403 404 // dump record threads 405 for (size_t i = 0; i < mRecordThreads.size(); i++) { 406 mRecordThreads.valueAt(i)->dump(fd, args); 407 } 408 409 // dump all hardware devs 410 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 411 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 412 dev->dump(dev, fd); 413 } 414 if (locked) mLock.unlock(); 415 } 416 return NO_ERROR; 417} 418 419sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 420{ 421 // If pid is already in the mClients wp<> map, then use that entry 422 // (for which promote() is always != 0), otherwise create a new entry and Client. 423 sp<Client> client = mClients.valueFor(pid).promote(); 424 if (client == 0) { 425 client = new Client(this, pid); 426 mClients.add(pid, client); 427 } 428 429 return client; 430} 431 432// IAudioFlinger interface 433 434 435sp<IAudioTrack> AudioFlinger::createTrack( 436 pid_t pid, 437 audio_stream_type_t streamType, 438 uint32_t sampleRate, 439 audio_format_t format, 440 uint32_t channelMask, 441 int frameCount, 442 IAudioFlinger::track_flags_t flags, 443 const sp<IMemory>& sharedBuffer, 444 audio_io_handle_t output, 445 pid_t tid, 446 int *sessionId, 447 status_t *status) 448{ 449 sp<PlaybackThread::Track> track; 450 sp<TrackHandle> trackHandle; 451 sp<Client> client; 452 status_t lStatus; 453 int lSessionId; 454 455 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 456 // but if someone uses binder directly they could bypass that and cause us to crash 457 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 458 ALOGE("createTrack() invalid stream type %d", streamType); 459 lStatus = BAD_VALUE; 460 goto Exit; 461 } 462 463 { 464 Mutex::Autolock _l(mLock); 465 PlaybackThread *thread = checkPlaybackThread_l(output); 466 PlaybackThread *effectThread = NULL; 467 if (thread == NULL) { 468 ALOGE("unknown output thread"); 469 lStatus = BAD_VALUE; 470 goto Exit; 471 } 472 473 client = registerPid_l(pid); 474 475 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 476 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 477 // check if an effect chain with the same session ID is present on another 478 // output thread and move it here. 479 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 480 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 481 if (mPlaybackThreads.keyAt(i) != output) { 482 uint32_t sessions = t->hasAudioSession(*sessionId); 483 if (sessions & PlaybackThread::EFFECT_SESSION) { 484 effectThread = t.get(); 485 break; 486 } 487 } 488 } 489 lSessionId = *sessionId; 490 } else { 491 // if no audio session id is provided, create one here 492 lSessionId = nextUniqueId(); 493 if (sessionId != NULL) { 494 *sessionId = lSessionId; 495 } 496 } 497 ALOGV("createTrack() lSessionId: %d", lSessionId); 498 499 track = thread->createTrack_l(client, streamType, sampleRate, format, 500 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 501 502 // move effect chain to this output thread if an effect on same session was waiting 503 // for a track to be created 504 if (lStatus == NO_ERROR && effectThread != NULL) { 505 Mutex::Autolock _dl(thread->mLock); 506 Mutex::Autolock _sl(effectThread->mLock); 507 moveEffectChain_l(lSessionId, effectThread, thread, true); 508 } 509 510 // Look for sync events awaiting for a session to be used. 511 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 512 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 513 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 514 if (lStatus == NO_ERROR) { 515 track->setSyncEvent(mPendingSyncEvents[i]); 516 } else { 517 mPendingSyncEvents[i]->cancel(); 518 } 519 mPendingSyncEvents.removeAt(i); 520 i--; 521 } 522 } 523 } 524 } 525 if (lStatus == NO_ERROR) { 526 trackHandle = new TrackHandle(track); 527 } else { 528 // remove local strong reference to Client before deleting the Track so that the Client 529 // destructor is called by the TrackBase destructor with mLock held 530 client.clear(); 531 track.clear(); 532 } 533 534Exit: 535 if (status != NULL) { 536 *status = lStatus; 537 } 538 return trackHandle; 539} 540 541uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 542{ 543 Mutex::Autolock _l(mLock); 544 PlaybackThread *thread = checkPlaybackThread_l(output); 545 if (thread == NULL) { 546 ALOGW("sampleRate() unknown thread %d", output); 547 return 0; 548 } 549 return thread->sampleRate(); 550} 551 552int AudioFlinger::channelCount(audio_io_handle_t output) const 553{ 554 Mutex::Autolock _l(mLock); 555 PlaybackThread *thread = checkPlaybackThread_l(output); 556 if (thread == NULL) { 557 ALOGW("channelCount() unknown thread %d", output); 558 return 0; 559 } 560 return thread->channelCount(); 561} 562 563audio_format_t AudioFlinger::format(audio_io_handle_t output) const 564{ 565 Mutex::Autolock _l(mLock); 566 PlaybackThread *thread = checkPlaybackThread_l(output); 567 if (thread == NULL) { 568 ALOGW("format() unknown thread %d", output); 569 return AUDIO_FORMAT_INVALID; 570 } 571 return thread->format(); 572} 573 574size_t AudioFlinger::frameCount(audio_io_handle_t output) const 575{ 576 Mutex::Autolock _l(mLock); 577 PlaybackThread *thread = checkPlaybackThread_l(output); 578 if (thread == NULL) { 579 ALOGW("frameCount() unknown thread %d", output); 580 return 0; 581 } 582 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 583 // should examine all callers and fix them to handle smaller counts 584 return thread->frameCount(); 585} 586 587uint32_t AudioFlinger::latency(audio_io_handle_t output) const 588{ 589 Mutex::Autolock _l(mLock); 590 PlaybackThread *thread = checkPlaybackThread_l(output); 591 if (thread == NULL) { 592 ALOGW("latency() unknown thread %d", output); 593 return 0; 594 } 595 return thread->latency(); 596} 597 598status_t AudioFlinger::setMasterVolume(float value) 599{ 600 status_t ret = initCheck(); 601 if (ret != NO_ERROR) { 602 return ret; 603 } 604 605 // check calling permissions 606 if (!settingsAllowed()) { 607 return PERMISSION_DENIED; 608 } 609 610 float swmv = value; 611 612 Mutex::Autolock _l(mLock); 613 614 // when hw supports master volume, don't scale in sw mixer 615 if (MVS_NONE != mMasterVolumeSupportLvl) { 616 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 617 AutoMutex lock(mHardwareLock); 618 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 619 620 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 621 if (NULL != dev->set_master_volume) { 622 dev->set_master_volume(dev, value); 623 } 624 mHardwareStatus = AUDIO_HW_IDLE; 625 } 626 627 swmv = 1.0; 628 } 629 630 mMasterVolume = value; 631 mMasterVolumeSW = swmv; 632 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 633 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 634 635 return NO_ERROR; 636} 637 638status_t AudioFlinger::setMode(audio_mode_t mode) 639{ 640 status_t ret = initCheck(); 641 if (ret != NO_ERROR) { 642 return ret; 643 } 644 645 // check calling permissions 646 if (!settingsAllowed()) { 647 return PERMISSION_DENIED; 648 } 649 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 650 ALOGW("Illegal value: setMode(%d)", mode); 651 return BAD_VALUE; 652 } 653 654 { // scope for the lock 655 AutoMutex lock(mHardwareLock); 656 mHardwareStatus = AUDIO_HW_SET_MODE; 657 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 658 mHardwareStatus = AUDIO_HW_IDLE; 659 } 660 661 if (NO_ERROR == ret) { 662 Mutex::Autolock _l(mLock); 663 mMode = mode; 664 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 665 mPlaybackThreads.valueAt(i)->setMode(mode); 666 } 667 668 return ret; 669} 670 671status_t AudioFlinger::setMicMute(bool state) 672{ 673 status_t ret = initCheck(); 674 if (ret != NO_ERROR) { 675 return ret; 676 } 677 678 // check calling permissions 679 if (!settingsAllowed()) { 680 return PERMISSION_DENIED; 681 } 682 683 AutoMutex lock(mHardwareLock); 684 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 685 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 686 mHardwareStatus = AUDIO_HW_IDLE; 687 return ret; 688} 689 690bool AudioFlinger::getMicMute() const 691{ 692 status_t ret = initCheck(); 693 if (ret != NO_ERROR) { 694 return false; 695 } 696 697 bool state = AUDIO_MODE_INVALID; 698 AutoMutex lock(mHardwareLock); 699 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 700 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 701 mHardwareStatus = AUDIO_HW_IDLE; 702 return state; 703} 704 705status_t AudioFlinger::setMasterMute(bool muted) 706{ 707 // check calling permissions 708 if (!settingsAllowed()) { 709 return PERMISSION_DENIED; 710 } 711 712 Mutex::Autolock _l(mLock); 713 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 714 mMasterMute = muted; 715 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 716 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 717 718 return NO_ERROR; 719} 720 721float AudioFlinger::masterVolume() const 722{ 723 Mutex::Autolock _l(mLock); 724 return masterVolume_l(); 725} 726 727float AudioFlinger::masterVolumeSW() const 728{ 729 Mutex::Autolock _l(mLock); 730 return masterVolumeSW_l(); 731} 732 733bool AudioFlinger::masterMute() const 734{ 735 Mutex::Autolock _l(mLock); 736 return masterMute_l(); 737} 738 739float AudioFlinger::masterVolume_l() const 740{ 741 if (MVS_FULL == mMasterVolumeSupportLvl) { 742 float ret_val; 743 AutoMutex lock(mHardwareLock); 744 745 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 746 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 747 (NULL != mPrimaryHardwareDev->get_master_volume), 748 "can't get master volume"); 749 750 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 751 mHardwareStatus = AUDIO_HW_IDLE; 752 return ret_val; 753 } 754 755 return mMasterVolume; 756} 757 758status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 759 audio_io_handle_t output) 760{ 761 // check calling permissions 762 if (!settingsAllowed()) { 763 return PERMISSION_DENIED; 764 } 765 766 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 767 ALOGE("setStreamVolume() invalid stream %d", stream); 768 return BAD_VALUE; 769 } 770 771 AutoMutex lock(mLock); 772 PlaybackThread *thread = NULL; 773 if (output) { 774 thread = checkPlaybackThread_l(output); 775 if (thread == NULL) { 776 return BAD_VALUE; 777 } 778 } 779 780 mStreamTypes[stream].volume = value; 781 782 if (thread == NULL) { 783 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 784 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 785 } 786 } else { 787 thread->setStreamVolume(stream, value); 788 } 789 790 return NO_ERROR; 791} 792 793status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 794{ 795 // check calling permissions 796 if (!settingsAllowed()) { 797 return PERMISSION_DENIED; 798 } 799 800 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 801 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 802 ALOGE("setStreamMute() invalid stream %d", stream); 803 return BAD_VALUE; 804 } 805 806 AutoMutex lock(mLock); 807 mStreamTypes[stream].mute = muted; 808 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 809 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 810 811 return NO_ERROR; 812} 813 814float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return 0.0f; 818 } 819 820 AutoMutex lock(mLock); 821 float volume; 822 if (output) { 823 PlaybackThread *thread = checkPlaybackThread_l(output); 824 if (thread == NULL) { 825 return 0.0f; 826 } 827 volume = thread->streamVolume(stream); 828 } else { 829 volume = streamVolume_l(stream); 830 } 831 832 return volume; 833} 834 835bool AudioFlinger::streamMute(audio_stream_type_t stream) const 836{ 837 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 838 return true; 839 } 840 841 AutoMutex lock(mLock); 842 return streamMute_l(stream); 843} 844 845status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 846{ 847 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 848 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 849 // check calling permissions 850 if (!settingsAllowed()) { 851 return PERMISSION_DENIED; 852 } 853 854 // ioHandle == 0 means the parameters are global to the audio hardware interface 855 if (ioHandle == 0) { 856 Mutex::Autolock _l(mLock); 857 status_t final_result = NO_ERROR; 858 { 859 AutoMutex lock(mHardwareLock); 860 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 861 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 862 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 863 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 864 final_result = result ?: final_result; 865 } 866 mHardwareStatus = AUDIO_HW_IDLE; 867 } 868 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 869 AudioParameter param = AudioParameter(keyValuePairs); 870 String8 value; 871 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 872 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 873 if (mBtNrecIsOff != btNrecIsOff) { 874 for (size_t i = 0; i < mRecordThreads.size(); i++) { 875 sp<RecordThread> thread = mRecordThreads.valueAt(i); 876 RecordThread::RecordTrack *track = thread->track(); 877 if (track != NULL) { 878 audio_devices_t device = (audio_devices_t)( 879 thread->device() & AUDIO_DEVICE_IN_ALL); 880 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 881 thread->setEffectSuspended(FX_IID_AEC, 882 suspend, 883 track->sessionId()); 884 thread->setEffectSuspended(FX_IID_NS, 885 suspend, 886 track->sessionId()); 887 } 888 } 889 mBtNrecIsOff = btNrecIsOff; 890 } 891 } 892 return final_result; 893 } 894 895 // hold a strong ref on thread in case closeOutput() or closeInput() is called 896 // and the thread is exited once the lock is released 897 sp<ThreadBase> thread; 898 { 899 Mutex::Autolock _l(mLock); 900 thread = checkPlaybackThread_l(ioHandle); 901 if (thread == NULL) { 902 thread = checkRecordThread_l(ioHandle); 903 } else if (thread == primaryPlaybackThread_l()) { 904 // indicate output device change to all input threads for pre processing 905 AudioParameter param = AudioParameter(keyValuePairs); 906 int value; 907 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 908 (value != 0)) { 909 for (size_t i = 0; i < mRecordThreads.size(); i++) { 910 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 911 } 912 } 913 } 914 } 915 if (thread != 0) { 916 return thread->setParameters(keyValuePairs); 917 } 918 return BAD_VALUE; 919} 920 921String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 922{ 923// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 924// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 925 926 Mutex::Autolock _l(mLock); 927 928 if (ioHandle == 0) { 929 String8 out_s8; 930 931 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 932 char *s; 933 { 934 AutoMutex lock(mHardwareLock); 935 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 936 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 937 s = dev->get_parameters(dev, keys.string()); 938 mHardwareStatus = AUDIO_HW_IDLE; 939 } 940 out_s8 += String8(s ? s : ""); 941 free(s); 942 } 943 return out_s8; 944 } 945 946 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 947 if (playbackThread != NULL) { 948 return playbackThread->getParameters(keys); 949 } 950 RecordThread *recordThread = checkRecordThread_l(ioHandle); 951 if (recordThread != NULL) { 952 return recordThread->getParameters(keys); 953 } 954 return String8(""); 955} 956 957size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 958{ 959 status_t ret = initCheck(); 960 if (ret != NO_ERROR) { 961 return 0; 962 } 963 964 AutoMutex lock(mHardwareLock); 965 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 966 struct audio_config config = { 967 sample_rate: sampleRate, 968 channel_mask: audio_channel_in_mask_from_count(channelCount), 969 format: format, 970 }; 971 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 972 mHardwareStatus = AUDIO_HW_IDLE; 973 return size; 974} 975 976unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 977{ 978 if (ioHandle == 0) { 979 return 0; 980 } 981 982 Mutex::Autolock _l(mLock); 983 984 RecordThread *recordThread = checkRecordThread_l(ioHandle); 985 if (recordThread != NULL) { 986 return recordThread->getInputFramesLost(); 987 } 988 return 0; 989} 990 991status_t AudioFlinger::setVoiceVolume(float value) 992{ 993 status_t ret = initCheck(); 994 if (ret != NO_ERROR) { 995 return ret; 996 } 997 998 // check calling permissions 999 if (!settingsAllowed()) { 1000 return PERMISSION_DENIED; 1001 } 1002 1003 AutoMutex lock(mHardwareLock); 1004 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1005 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1006 mHardwareStatus = AUDIO_HW_IDLE; 1007 1008 return ret; 1009} 1010 1011status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1012 audio_io_handle_t output) const 1013{ 1014 status_t status; 1015 1016 Mutex::Autolock _l(mLock); 1017 1018 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1019 if (playbackThread != NULL) { 1020 return playbackThread->getRenderPosition(halFrames, dspFrames); 1021 } 1022 1023 return BAD_VALUE; 1024} 1025 1026void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1027{ 1028 1029 Mutex::Autolock _l(mLock); 1030 1031 pid_t pid = IPCThreadState::self()->getCallingPid(); 1032 if (mNotificationClients.indexOfKey(pid) < 0) { 1033 sp<NotificationClient> notificationClient = new NotificationClient(this, 1034 client, 1035 pid); 1036 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1037 1038 mNotificationClients.add(pid, notificationClient); 1039 1040 sp<IBinder> binder = client->asBinder(); 1041 binder->linkToDeath(notificationClient); 1042 1043 // the config change is always sent from playback or record threads to avoid deadlock 1044 // with AudioSystem::gLock 1045 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1046 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1047 } 1048 1049 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1050 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1051 } 1052 } 1053} 1054 1055void AudioFlinger::removeNotificationClient(pid_t pid) 1056{ 1057 Mutex::Autolock _l(mLock); 1058 1059 mNotificationClients.removeItem(pid); 1060 1061 ALOGV("%d died, releasing its sessions", pid); 1062 size_t num = mAudioSessionRefs.size(); 1063 bool removed = false; 1064 for (size_t i = 0; i< num; ) { 1065 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1066 ALOGV(" pid %d @ %d", ref->mPid, i); 1067 if (ref->mPid == pid) { 1068 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1069 mAudioSessionRefs.removeAt(i); 1070 delete ref; 1071 removed = true; 1072 num--; 1073 } else { 1074 i++; 1075 } 1076 } 1077 if (removed) { 1078 purgeStaleEffects_l(); 1079 } 1080} 1081 1082// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1083void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1084{ 1085 size_t size = mNotificationClients.size(); 1086 for (size_t i = 0; i < size; i++) { 1087 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1088 param2); 1089 } 1090} 1091 1092// removeClient_l() must be called with AudioFlinger::mLock held 1093void AudioFlinger::removeClient_l(pid_t pid) 1094{ 1095 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1096 mClients.removeItem(pid); 1097} 1098 1099 1100// ---------------------------------------------------------------------------- 1101 1102AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1103 uint32_t device, type_t type) 1104 : Thread(false), 1105 mType(type), 1106 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1107 // mChannelMask 1108 mChannelCount(0), 1109 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1110 mParamStatus(NO_ERROR), 1111 mStandby(false), mId(id), 1112 mDevice(device), 1113 mDeathRecipient(new PMDeathRecipient(this)) 1114{ 1115} 1116 1117AudioFlinger::ThreadBase::~ThreadBase() 1118{ 1119 mParamCond.broadcast(); 1120 // do not lock the mutex in destructor 1121 releaseWakeLock_l(); 1122 if (mPowerManager != 0) { 1123 sp<IBinder> binder = mPowerManager->asBinder(); 1124 binder->unlinkToDeath(mDeathRecipient); 1125 } 1126} 1127 1128void AudioFlinger::ThreadBase::exit() 1129{ 1130 ALOGV("ThreadBase::exit"); 1131 { 1132 // This lock prevents the following race in thread (uniprocessor for illustration): 1133 // if (!exitPending()) { 1134 // // context switch from here to exit() 1135 // // exit() calls requestExit(), what exitPending() observes 1136 // // exit() calls signal(), which is dropped since no waiters 1137 // // context switch back from exit() to here 1138 // mWaitWorkCV.wait(...); 1139 // // now thread is hung 1140 // } 1141 AutoMutex lock(mLock); 1142 requestExit(); 1143 mWaitWorkCV.signal(); 1144 } 1145 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1146 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1147 requestExitAndWait(); 1148} 1149 1150status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1151{ 1152 status_t status; 1153 1154 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1155 Mutex::Autolock _l(mLock); 1156 1157 mNewParameters.add(keyValuePairs); 1158 mWaitWorkCV.signal(); 1159 // wait condition with timeout in case the thread loop has exited 1160 // before the request could be processed 1161 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1162 status = mParamStatus; 1163 mWaitWorkCV.signal(); 1164 } else { 1165 status = TIMED_OUT; 1166 } 1167 return status; 1168} 1169 1170void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1171{ 1172 Mutex::Autolock _l(mLock); 1173 sendConfigEvent_l(event, param); 1174} 1175 1176// sendConfigEvent_l() must be called with ThreadBase::mLock held 1177void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1178{ 1179 ConfigEvent configEvent; 1180 configEvent.mEvent = event; 1181 configEvent.mParam = param; 1182 mConfigEvents.add(configEvent); 1183 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1184 mWaitWorkCV.signal(); 1185} 1186 1187void AudioFlinger::ThreadBase::processConfigEvents() 1188{ 1189 mLock.lock(); 1190 while (!mConfigEvents.isEmpty()) { 1191 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1192 ConfigEvent configEvent = mConfigEvents[0]; 1193 mConfigEvents.removeAt(0); 1194 // release mLock before locking AudioFlinger mLock: lock order is always 1195 // AudioFlinger then ThreadBase to avoid cross deadlock 1196 mLock.unlock(); 1197 mAudioFlinger->mLock.lock(); 1198 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1199 mAudioFlinger->mLock.unlock(); 1200 mLock.lock(); 1201 } 1202 mLock.unlock(); 1203} 1204 1205status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1206{ 1207 const size_t SIZE = 256; 1208 char buffer[SIZE]; 1209 String8 result; 1210 1211 bool locked = tryLock(mLock); 1212 if (!locked) { 1213 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1214 write(fd, buffer, strlen(buffer)); 1215 } 1216 1217 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1218 result.append(buffer); 1219 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1220 result.append(buffer); 1221 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1222 result.append(buffer); 1223 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1224 result.append(buffer); 1225 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1226 result.append(buffer); 1227 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1228 result.append(buffer); 1229 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1230 result.append(buffer); 1231 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1232 result.append(buffer); 1233 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1234 result.append(buffer); 1235 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1236 result.append(buffer); 1237 1238 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1239 result.append(buffer); 1240 result.append(" Index Command"); 1241 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1242 snprintf(buffer, SIZE, "\n %02d ", i); 1243 result.append(buffer); 1244 result.append(mNewParameters[i]); 1245 } 1246 1247 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1248 result.append(buffer); 1249 snprintf(buffer, SIZE, " Index event param\n"); 1250 result.append(buffer); 1251 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1252 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1253 result.append(buffer); 1254 } 1255 result.append("\n"); 1256 1257 write(fd, result.string(), result.size()); 1258 1259 if (locked) { 1260 mLock.unlock(); 1261 } 1262 return NO_ERROR; 1263} 1264 1265status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1266{ 1267 const size_t SIZE = 256; 1268 char buffer[SIZE]; 1269 String8 result; 1270 1271 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1272 write(fd, buffer, strlen(buffer)); 1273 1274 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1275 sp<EffectChain> chain = mEffectChains[i]; 1276 if (chain != 0) { 1277 chain->dump(fd, args); 1278 } 1279 } 1280 return NO_ERROR; 1281} 1282 1283void AudioFlinger::ThreadBase::acquireWakeLock() 1284{ 1285 Mutex::Autolock _l(mLock); 1286 acquireWakeLock_l(); 1287} 1288 1289void AudioFlinger::ThreadBase::acquireWakeLock_l() 1290{ 1291 if (mPowerManager == 0) { 1292 // use checkService() to avoid blocking if power service is not up yet 1293 sp<IBinder> binder = 1294 defaultServiceManager()->checkService(String16("power")); 1295 if (binder == 0) { 1296 ALOGW("Thread %s cannot connect to the power manager service", mName); 1297 } else { 1298 mPowerManager = interface_cast<IPowerManager>(binder); 1299 binder->linkToDeath(mDeathRecipient); 1300 } 1301 } 1302 if (mPowerManager != 0) { 1303 sp<IBinder> binder = new BBinder(); 1304 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1305 binder, 1306 String16(mName)); 1307 if (status == NO_ERROR) { 1308 mWakeLockToken = binder; 1309 } 1310 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1311 } 1312} 1313 1314void AudioFlinger::ThreadBase::releaseWakeLock() 1315{ 1316 Mutex::Autolock _l(mLock); 1317 releaseWakeLock_l(); 1318} 1319 1320void AudioFlinger::ThreadBase::releaseWakeLock_l() 1321{ 1322 if (mWakeLockToken != 0) { 1323 ALOGV("releaseWakeLock_l() %s", mName); 1324 if (mPowerManager != 0) { 1325 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1326 } 1327 mWakeLockToken.clear(); 1328 } 1329} 1330 1331void AudioFlinger::ThreadBase::clearPowerManager() 1332{ 1333 Mutex::Autolock _l(mLock); 1334 releaseWakeLock_l(); 1335 mPowerManager.clear(); 1336} 1337 1338void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1339{ 1340 sp<ThreadBase> thread = mThread.promote(); 1341 if (thread != 0) { 1342 thread->clearPowerManager(); 1343 } 1344 ALOGW("power manager service died !!!"); 1345} 1346 1347void AudioFlinger::ThreadBase::setEffectSuspended( 1348 const effect_uuid_t *type, bool suspend, int sessionId) 1349{ 1350 Mutex::Autolock _l(mLock); 1351 setEffectSuspended_l(type, suspend, sessionId); 1352} 1353 1354void AudioFlinger::ThreadBase::setEffectSuspended_l( 1355 const effect_uuid_t *type, bool suspend, int sessionId) 1356{ 1357 sp<EffectChain> chain = getEffectChain_l(sessionId); 1358 if (chain != 0) { 1359 if (type != NULL) { 1360 chain->setEffectSuspended_l(type, suspend); 1361 } else { 1362 chain->setEffectSuspendedAll_l(suspend); 1363 } 1364 } 1365 1366 updateSuspendedSessions_l(type, suspend, sessionId); 1367} 1368 1369void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1370{ 1371 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1372 if (index < 0) { 1373 return; 1374 } 1375 1376 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1377 mSuspendedSessions.editValueAt(index); 1378 1379 for (size_t i = 0; i < sessionEffects.size(); i++) { 1380 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1381 for (int j = 0; j < desc->mRefCount; j++) { 1382 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1383 chain->setEffectSuspendedAll_l(true); 1384 } else { 1385 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1386 desc->mType.timeLow); 1387 chain->setEffectSuspended_l(&desc->mType, true); 1388 } 1389 } 1390 } 1391} 1392 1393void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1394 bool suspend, 1395 int sessionId) 1396{ 1397 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1398 1399 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1400 1401 if (suspend) { 1402 if (index >= 0) { 1403 sessionEffects = mSuspendedSessions.editValueAt(index); 1404 } else { 1405 mSuspendedSessions.add(sessionId, sessionEffects); 1406 } 1407 } else { 1408 if (index < 0) { 1409 return; 1410 } 1411 sessionEffects = mSuspendedSessions.editValueAt(index); 1412 } 1413 1414 1415 int key = EffectChain::kKeyForSuspendAll; 1416 if (type != NULL) { 1417 key = type->timeLow; 1418 } 1419 index = sessionEffects.indexOfKey(key); 1420 1421 sp<SuspendedSessionDesc> desc; 1422 if (suspend) { 1423 if (index >= 0) { 1424 desc = sessionEffects.valueAt(index); 1425 } else { 1426 desc = new SuspendedSessionDesc(); 1427 if (type != NULL) { 1428 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1429 } 1430 sessionEffects.add(key, desc); 1431 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1432 } 1433 desc->mRefCount++; 1434 } else { 1435 if (index < 0) { 1436 return; 1437 } 1438 desc = sessionEffects.valueAt(index); 1439 if (--desc->mRefCount == 0) { 1440 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1441 sessionEffects.removeItemsAt(index); 1442 if (sessionEffects.isEmpty()) { 1443 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1444 sessionId); 1445 mSuspendedSessions.removeItem(sessionId); 1446 } 1447 } 1448 } 1449 if (!sessionEffects.isEmpty()) { 1450 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1451 } 1452} 1453 1454void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1455 bool enabled, 1456 int sessionId) 1457{ 1458 Mutex::Autolock _l(mLock); 1459 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1460} 1461 1462void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1463 bool enabled, 1464 int sessionId) 1465{ 1466 if (mType != RECORD) { 1467 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1468 // another session. This gives the priority to well behaved effect control panels 1469 // and applications not using global effects. 1470 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1471 // global effects 1472 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1473 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1474 } 1475 } 1476 1477 sp<EffectChain> chain = getEffectChain_l(sessionId); 1478 if (chain != 0) { 1479 chain->checkSuspendOnEffectEnabled(effect, enabled); 1480 } 1481} 1482 1483// ---------------------------------------------------------------------------- 1484 1485AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1486 AudioStreamOut* output, 1487 audio_io_handle_t id, 1488 uint32_t device, 1489 type_t type) 1490 : ThreadBase(audioFlinger, id, device, type), 1491 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1492 // Assumes constructor is called by AudioFlinger with it's mLock held, 1493 // but it would be safer to explicitly pass initial masterMute as parameter 1494 mMasterMute(audioFlinger->masterMute_l()), 1495 // mStreamTypes[] initialized in constructor body 1496 mOutput(output), 1497 // Assumes constructor is called by AudioFlinger with it's mLock held, 1498 // but it would be safer to explicitly pass initial masterVolume as parameter 1499 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1500 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1501 mMixerStatus(MIXER_IDLE), 1502 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1503 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1504 // index 0 is reserved for normal mixer's submix 1505 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1506{ 1507 snprintf(mName, kNameLength, "AudioOut_%X", id); 1508 1509 readOutputParameters(); 1510 1511 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1512 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1513 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1514 stream = (audio_stream_type_t) (stream + 1)) { 1515 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1516 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1517 } 1518 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1519 // because mAudioFlinger doesn't have one to copy from 1520} 1521 1522AudioFlinger::PlaybackThread::~PlaybackThread() 1523{ 1524 delete [] mMixBuffer; 1525} 1526 1527status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1528{ 1529 dumpInternals(fd, args); 1530 dumpTracks(fd, args); 1531 dumpEffectChains(fd, args); 1532 return NO_ERROR; 1533} 1534 1535status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1536{ 1537 const size_t SIZE = 256; 1538 char buffer[SIZE]; 1539 String8 result; 1540 1541 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1542 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1543 const stream_type_t *st = &mStreamTypes[i]; 1544 if (i > 0) { 1545 result.appendFormat(", "); 1546 } 1547 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1548 if (st->mute) { 1549 result.append("M"); 1550 } 1551 } 1552 result.append("\n"); 1553 write(fd, result.string(), result.length()); 1554 result.clear(); 1555 1556 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1557 result.append(buffer); 1558 Track::appendDumpHeader(result); 1559 for (size_t i = 0; i < mTracks.size(); ++i) { 1560 sp<Track> track = mTracks[i]; 1561 if (track != 0) { 1562 track->dump(buffer, SIZE); 1563 result.append(buffer); 1564 } 1565 } 1566 1567 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1568 result.append(buffer); 1569 Track::appendDumpHeader(result); 1570 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1571 sp<Track> track = mActiveTracks[i].promote(); 1572 if (track != 0) { 1573 track->dump(buffer, SIZE); 1574 result.append(buffer); 1575 } 1576 } 1577 write(fd, result.string(), result.size()); 1578 1579 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1580 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1581 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1582 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1583 1584 return NO_ERROR; 1585} 1586 1587status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1588{ 1589 const size_t SIZE = 256; 1590 char buffer[SIZE]; 1591 String8 result; 1592 1593 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1594 result.append(buffer); 1595 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1596 result.append(buffer); 1597 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1598 result.append(buffer); 1599 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1600 result.append(buffer); 1601 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1602 result.append(buffer); 1603 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1604 result.append(buffer); 1605 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1606 result.append(buffer); 1607 write(fd, result.string(), result.size()); 1608 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1609 1610 dumpBase(fd, args); 1611 1612 return NO_ERROR; 1613} 1614 1615// Thread virtuals 1616status_t AudioFlinger::PlaybackThread::readyToRun() 1617{ 1618 status_t status = initCheck(); 1619 if (status == NO_ERROR) { 1620 ALOGI("AudioFlinger's thread %p ready to run", this); 1621 } else { 1622 ALOGE("No working audio driver found."); 1623 } 1624 return status; 1625} 1626 1627void AudioFlinger::PlaybackThread::onFirstRef() 1628{ 1629 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1630} 1631 1632// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1633sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1634 const sp<AudioFlinger::Client>& client, 1635 audio_stream_type_t streamType, 1636 uint32_t sampleRate, 1637 audio_format_t format, 1638 uint32_t channelMask, 1639 int frameCount, 1640 const sp<IMemory>& sharedBuffer, 1641 int sessionId, 1642 IAudioFlinger::track_flags_t flags, 1643 pid_t tid, 1644 status_t *status) 1645{ 1646 sp<Track> track; 1647 status_t lStatus; 1648 1649 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1650 1651 // client expresses a preference for FAST, but we get the final say 1652 if (flags & IAudioFlinger::TRACK_FAST) { 1653 if ( 1654 // not timed 1655 (!isTimed) && 1656 // either of these use cases: 1657 ( 1658 // use case 1: shared buffer with any frame count 1659 ( 1660 (sharedBuffer != 0) 1661 ) || 1662 // use case 2: callback handler and frame count is default or at least as large as HAL 1663 ( 1664 (tid != -1) && 1665 ((frameCount == 0) || 1666 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1667 ) 1668 ) && 1669 // PCM data 1670 audio_is_linear_pcm(format) && 1671 // mono or stereo 1672 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1673 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1674#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1675 // hardware sample rate 1676 (sampleRate == mSampleRate) && 1677#endif 1678 // normal mixer has an associated fast mixer 1679 hasFastMixer() && 1680 // there are sufficient fast track slots available 1681 (mFastTrackAvailMask != 0) 1682 // FIXME test that MixerThread for this fast track has a capable output HAL 1683 // FIXME add a permission test also? 1684 ) { 1685 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1686 if (frameCount == 0) { 1687 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1688 } 1689 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1690 frameCount, mFrameCount); 1691 } else { 1692 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1693 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1694 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1695 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1696 audio_is_linear_pcm(format), 1697 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1698 flags &= ~IAudioFlinger::TRACK_FAST; 1699 // For compatibility with AudioTrack calculation, buffer depth is forced 1700 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1701 // This is probably too conservative, but legacy application code may depend on it. 1702 // If you change this calculation, also review the start threshold which is related. 1703 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1704 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1705 if (minBufCount < 2) { 1706 minBufCount = 2; 1707 } 1708 int minFrameCount = mNormalFrameCount * minBufCount; 1709 if (frameCount < minFrameCount) { 1710 frameCount = minFrameCount; 1711 } 1712 } 1713 } 1714 1715 if (mType == DIRECT) { 1716 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1717 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1718 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1719 "for output %p with format %d", 1720 sampleRate, format, channelMask, mOutput, mFormat); 1721 lStatus = BAD_VALUE; 1722 goto Exit; 1723 } 1724 } 1725 } else { 1726 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1727 if (sampleRate > mSampleRate*2) { 1728 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1729 lStatus = BAD_VALUE; 1730 goto Exit; 1731 } 1732 } 1733 1734 lStatus = initCheck(); 1735 if (lStatus != NO_ERROR) { 1736 ALOGE("Audio driver not initialized."); 1737 goto Exit; 1738 } 1739 1740 { // scope for mLock 1741 Mutex::Autolock _l(mLock); 1742 1743 // all tracks in same audio session must share the same routing strategy otherwise 1744 // conflicts will happen when tracks are moved from one output to another by audio policy 1745 // manager 1746 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1747 for (size_t i = 0; i < mTracks.size(); ++i) { 1748 sp<Track> t = mTracks[i]; 1749 if (t != 0 && !t->isOutputTrack()) { 1750 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1751 if (sessionId == t->sessionId() && strategy != actual) { 1752 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1753 strategy, actual); 1754 lStatus = BAD_VALUE; 1755 goto Exit; 1756 } 1757 } 1758 } 1759 1760 if (!isTimed) { 1761 track = new Track(this, client, streamType, sampleRate, format, 1762 channelMask, frameCount, sharedBuffer, sessionId, flags); 1763 } else { 1764 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1765 channelMask, frameCount, sharedBuffer, sessionId); 1766 } 1767 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1768 lStatus = NO_MEMORY; 1769 goto Exit; 1770 } 1771 mTracks.add(track); 1772 1773 sp<EffectChain> chain = getEffectChain_l(sessionId); 1774 if (chain != 0) { 1775 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1776 track->setMainBuffer(chain->inBuffer()); 1777 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1778 chain->incTrackCnt(); 1779 } 1780 } 1781 1782#ifdef HAVE_REQUEST_PRIORITY 1783 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1784 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1785 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1786 // so ask activity manager to do this on our behalf 1787 int err = requestPriority(callingPid, tid, 1); 1788 if (err != 0) { 1789 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1790 1, callingPid, tid, err); 1791 } 1792 } 1793#endif 1794 1795 lStatus = NO_ERROR; 1796 1797Exit: 1798 if (status) { 1799 *status = lStatus; 1800 } 1801 return track; 1802} 1803 1804uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1805{ 1806 if (mFastMixer != NULL) { 1807 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1808 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1809 } 1810 return latency; 1811} 1812 1813uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1814{ 1815 return latency; 1816} 1817 1818uint32_t AudioFlinger::PlaybackThread::latency() const 1819{ 1820 Mutex::Autolock _l(mLock); 1821 if (initCheck() == NO_ERROR) { 1822 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1823 } else { 1824 return 0; 1825 } 1826} 1827 1828void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1829{ 1830 Mutex::Autolock _l(mLock); 1831 mMasterVolume = value; 1832} 1833 1834void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1835{ 1836 Mutex::Autolock _l(mLock); 1837 setMasterMute_l(muted); 1838} 1839 1840void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1841{ 1842 Mutex::Autolock _l(mLock); 1843 mStreamTypes[stream].volume = value; 1844} 1845 1846void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1847{ 1848 Mutex::Autolock _l(mLock); 1849 mStreamTypes[stream].mute = muted; 1850} 1851 1852float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1853{ 1854 Mutex::Autolock _l(mLock); 1855 return mStreamTypes[stream].volume; 1856} 1857 1858// addTrack_l() must be called with ThreadBase::mLock held 1859status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1860{ 1861 status_t status = ALREADY_EXISTS; 1862 1863 // set retry count for buffer fill 1864 track->mRetryCount = kMaxTrackStartupRetries; 1865 if (mActiveTracks.indexOf(track) < 0) { 1866 // the track is newly added, make sure it fills up all its 1867 // buffers before playing. This is to ensure the client will 1868 // effectively get the latency it requested. 1869 track->mFillingUpStatus = Track::FS_FILLING; 1870 track->mResetDone = false; 1871 track->mPresentationCompleteFrames = 0; 1872 mActiveTracks.add(track); 1873 if (track->mainBuffer() != mMixBuffer) { 1874 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1875 if (chain != 0) { 1876 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1877 chain->incActiveTrackCnt(); 1878 } 1879 } 1880 1881 status = NO_ERROR; 1882 } 1883 1884 ALOGV("mWaitWorkCV.broadcast"); 1885 mWaitWorkCV.broadcast(); 1886 1887 return status; 1888} 1889 1890// destroyTrack_l() must be called with ThreadBase::mLock held 1891void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1892{ 1893 track->mState = TrackBase::TERMINATED; 1894 // active tracks are removed by threadLoop() 1895 if (mActiveTracks.indexOf(track) < 0) { 1896 removeTrack_l(track); 1897 } 1898} 1899 1900void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1901{ 1902 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1903 mTracks.remove(track); 1904 deleteTrackName_l(track->name()); 1905 // redundant as track is about to be destroyed, for dumpsys only 1906 track->mName = -1; 1907 if (track->isFastTrack()) { 1908 int index = track->mFastIndex; 1909 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1910 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1911 mFastTrackAvailMask |= 1 << index; 1912 // redundant as track is about to be destroyed, for dumpsys only 1913 track->mFastIndex = -1; 1914 } 1915 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1916 if (chain != 0) { 1917 chain->decTrackCnt(); 1918 } 1919} 1920 1921String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1922{ 1923 String8 out_s8 = String8(""); 1924 char *s; 1925 1926 Mutex::Autolock _l(mLock); 1927 if (initCheck() != NO_ERROR) { 1928 return out_s8; 1929 } 1930 1931 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1932 out_s8 = String8(s); 1933 free(s); 1934 return out_s8; 1935} 1936 1937// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1938void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1939 AudioSystem::OutputDescriptor desc; 1940 void *param2 = NULL; 1941 1942 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1943 1944 switch (event) { 1945 case AudioSystem::OUTPUT_OPENED: 1946 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1947 desc.channels = mChannelMask; 1948 desc.samplingRate = mSampleRate; 1949 desc.format = mFormat; 1950 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1951 desc.latency = latency(); 1952 param2 = &desc; 1953 break; 1954 1955 case AudioSystem::STREAM_CONFIG_CHANGED: 1956 param2 = ¶m; 1957 case AudioSystem::OUTPUT_CLOSED: 1958 default: 1959 break; 1960 } 1961 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1962} 1963 1964void AudioFlinger::PlaybackThread::readOutputParameters() 1965{ 1966 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1967 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1968 mChannelCount = (uint16_t)popcount(mChannelMask); 1969 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1970 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1971 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1972 if (mFrameCount & 15) { 1973 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1974 mFrameCount); 1975 } 1976 1977 // Calculate size of normal mix buffer relative to the HAL output buffer size 1978 double multiplier = 1.0; 1979 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1980 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1981 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1982 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1983 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1984 maxNormalFrameCount = maxNormalFrameCount & ~15; 1985 if (maxNormalFrameCount < minNormalFrameCount) { 1986 maxNormalFrameCount = minNormalFrameCount; 1987 } 1988 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1989 if (multiplier <= 1.0) { 1990 multiplier = 1.0; 1991 } else if (multiplier <= 2.0) { 1992 if (2 * mFrameCount <= maxNormalFrameCount) { 1993 multiplier = 2.0; 1994 } else { 1995 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1996 } 1997 } else { 1998 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 1999 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2000 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2001 // FIXME this rounding up should not be done if no HAL SRC 2002 uint32_t truncMult = (uint32_t) multiplier; 2003 if ((truncMult & 1)) { 2004 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2005 ++truncMult; 2006 } 2007 } 2008 multiplier = (double) truncMult; 2009 } 2010 } 2011 mNormalFrameCount = multiplier * mFrameCount; 2012 // round up to nearest 16 frames to satisfy AudioMixer 2013 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2014 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2015 2016 delete[] mMixBuffer; 2017 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2018 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2019 2020 // force reconfiguration of effect chains and engines to take new buffer size and audio 2021 // parameters into account 2022 // Note that mLock is not held when readOutputParameters() is called from the constructor 2023 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2024 // matter. 2025 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2026 Vector< sp<EffectChain> > effectChains = mEffectChains; 2027 for (size_t i = 0; i < effectChains.size(); i ++) { 2028 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2029 } 2030} 2031 2032 2033status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2034{ 2035 if (halFrames == NULL || dspFrames == NULL) { 2036 return BAD_VALUE; 2037 } 2038 Mutex::Autolock _l(mLock); 2039 if (initCheck() != NO_ERROR) { 2040 return INVALID_OPERATION; 2041 } 2042 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2043 2044 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2045} 2046 2047uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2048{ 2049 Mutex::Autolock _l(mLock); 2050 uint32_t result = 0; 2051 if (getEffectChain_l(sessionId) != 0) { 2052 result = EFFECT_SESSION; 2053 } 2054 2055 for (size_t i = 0; i < mTracks.size(); ++i) { 2056 sp<Track> track = mTracks[i]; 2057 if (sessionId == track->sessionId() && 2058 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2059 result |= TRACK_SESSION; 2060 break; 2061 } 2062 } 2063 2064 return result; 2065} 2066 2067uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2068{ 2069 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2070 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2071 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2072 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2073 } 2074 for (size_t i = 0; i < mTracks.size(); i++) { 2075 sp<Track> track = mTracks[i]; 2076 if (sessionId == track->sessionId() && 2077 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2078 return AudioSystem::getStrategyForStream(track->streamType()); 2079 } 2080 } 2081 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2082} 2083 2084 2085AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2086{ 2087 Mutex::Autolock _l(mLock); 2088 return mOutput; 2089} 2090 2091AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2092{ 2093 Mutex::Autolock _l(mLock); 2094 AudioStreamOut *output = mOutput; 2095 mOutput = NULL; 2096 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2097 // must push a NULL and wait for ack 2098 mOutputSink.clear(); 2099 mPipeSink.clear(); 2100 mNormalSink.clear(); 2101 return output; 2102} 2103 2104// this method must always be called either with ThreadBase mLock held or inside the thread loop 2105audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2106{ 2107 if (mOutput == NULL) { 2108 return NULL; 2109 } 2110 return &mOutput->stream->common; 2111} 2112 2113uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2114{ 2115 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2116} 2117 2118status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2119{ 2120 if (!isValidSyncEvent(event)) { 2121 return BAD_VALUE; 2122 } 2123 2124 Mutex::Autolock _l(mLock); 2125 2126 for (size_t i = 0; i < mTracks.size(); ++i) { 2127 sp<Track> track = mTracks[i]; 2128 if (event->triggerSession() == track->sessionId()) { 2129 track->setSyncEvent(event); 2130 return NO_ERROR; 2131 } 2132 } 2133 2134 return NAME_NOT_FOUND; 2135} 2136 2137bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2138{ 2139 switch (event->type()) { 2140 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2141 return true; 2142 default: 2143 break; 2144 } 2145 return false; 2146} 2147 2148void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2149{ 2150 size_t count = tracksToRemove.size(); 2151 if (CC_UNLIKELY(count)) { 2152 for (size_t i = 0 ; i < count ; i++) { 2153 const sp<Track>& track = tracksToRemove.itemAt(i); 2154 if ((track->sharedBuffer() != 0) && 2155 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2156 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2157 } 2158 } 2159 } 2160 2161} 2162 2163// ---------------------------------------------------------------------------- 2164 2165AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2166 audio_io_handle_t id, uint32_t device, type_t type) 2167 : PlaybackThread(audioFlinger, output, id, device, type), 2168 // mAudioMixer below 2169#ifdef SOAKER 2170 mSoaker(NULL), 2171#endif 2172 // mFastMixer below 2173 mFastMixerFutex(0) 2174 // mOutputSink below 2175 // mPipeSink below 2176 // mNormalSink below 2177{ 2178 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2179 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2180 "mFrameCount=%d, mNormalFrameCount=%d", 2181 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2182 mNormalFrameCount); 2183 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2184 2185 // FIXME - Current mixer implementation only supports stereo output 2186 if (mChannelCount == 1) { 2187 ALOGE("Invalid audio hardware channel count"); 2188 } 2189 2190 // create an NBAIO sink for the HAL output stream, and negotiate 2191 mOutputSink = new AudioStreamOutSink(output->stream); 2192 size_t numCounterOffers = 0; 2193 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2194 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2195 ALOG_ASSERT(index == 0); 2196 2197 // initialize fast mixer depending on configuration 2198 bool initFastMixer; 2199 switch (kUseFastMixer) { 2200 case FastMixer_Never: 2201 initFastMixer = false; 2202 break; 2203 case FastMixer_Always: 2204 initFastMixer = true; 2205 break; 2206 case FastMixer_Static: 2207 case FastMixer_Dynamic: 2208 initFastMixer = mFrameCount < mNormalFrameCount; 2209 break; 2210 } 2211 if (initFastMixer) { 2212 2213 // create a MonoPipe to connect our submix to FastMixer 2214 NBAIO_Format format = mOutputSink->format(); 2215 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2216 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2217 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2218 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2219 const NBAIO_Format offers[1] = {format}; 2220 size_t numCounterOffers = 0; 2221 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2222 ALOG_ASSERT(index == 0); 2223 mPipeSink = monoPipe; 2224 2225#ifdef TEE_SINK_FRAMES 2226 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2227 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2228 numCounterOffers = 0; 2229 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2230 ALOG_ASSERT(index == 0); 2231 mTeeSink = teeSink; 2232 PipeReader *teeSource = new PipeReader(*teeSink); 2233 numCounterOffers = 0; 2234 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2235 ALOG_ASSERT(index == 0); 2236 mTeeSource = teeSource; 2237#endif 2238 2239#ifdef SOAKER 2240 // create a soaker as workaround for governor issues 2241 mSoaker = new Soaker(); 2242 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2243 mSoaker->run("Soaker", PRIORITY_LOWEST); 2244#endif 2245 2246 // create fast mixer and configure it initially with just one fast track for our submix 2247 mFastMixer = new FastMixer(); 2248 FastMixerStateQueue *sq = mFastMixer->sq(); 2249#ifdef STATE_QUEUE_DUMP 2250 sq->setObserverDump(&mStateQueueObserverDump); 2251 sq->setMutatorDump(&mStateQueueMutatorDump); 2252#endif 2253 FastMixerState *state = sq->begin(); 2254 FastTrack *fastTrack = &state->mFastTracks[0]; 2255 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2256 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2257 fastTrack->mVolumeProvider = NULL; 2258 fastTrack->mGeneration++; 2259 state->mFastTracksGen++; 2260 state->mTrackMask = 1; 2261 // fast mixer will use the HAL output sink 2262 state->mOutputSink = mOutputSink.get(); 2263 state->mOutputSinkGen++; 2264 state->mFrameCount = mFrameCount; 2265 state->mCommand = FastMixerState::COLD_IDLE; 2266 // already done in constructor initialization list 2267 //mFastMixerFutex = 0; 2268 state->mColdFutexAddr = &mFastMixerFutex; 2269 state->mColdGen++; 2270 state->mDumpState = &mFastMixerDumpState; 2271 state->mTeeSink = mTeeSink.get(); 2272 sq->end(); 2273 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2274 2275 // start the fast mixer 2276 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2277#ifdef HAVE_REQUEST_PRIORITY 2278 pid_t tid = mFastMixer->getTid(); 2279 int err = requestPriority(getpid_cached, tid, 2); 2280 if (err != 0) { 2281 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2282 2, getpid_cached, tid, err); 2283 } 2284#endif 2285 2286 } else { 2287 mFastMixer = NULL; 2288 } 2289 2290 switch (kUseFastMixer) { 2291 case FastMixer_Never: 2292 case FastMixer_Dynamic: 2293 mNormalSink = mOutputSink; 2294 break; 2295 case FastMixer_Always: 2296 mNormalSink = mPipeSink; 2297 break; 2298 case FastMixer_Static: 2299 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2300 break; 2301 } 2302} 2303 2304AudioFlinger::MixerThread::~MixerThread() 2305{ 2306 if (mFastMixer != NULL) { 2307 FastMixerStateQueue *sq = mFastMixer->sq(); 2308 FastMixerState *state = sq->begin(); 2309 if (state->mCommand == FastMixerState::COLD_IDLE) { 2310 int32_t old = android_atomic_inc(&mFastMixerFutex); 2311 if (old == -1) { 2312 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2313 } 2314 } 2315 state->mCommand = FastMixerState::EXIT; 2316 sq->end(); 2317 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2318 mFastMixer->join(); 2319 // Though the fast mixer thread has exited, it's state queue is still valid. 2320 // We'll use that extract the final state which contains one remaining fast track 2321 // corresponding to our sub-mix. 2322 state = sq->begin(); 2323 ALOG_ASSERT(state->mTrackMask == 1); 2324 FastTrack *fastTrack = &state->mFastTracks[0]; 2325 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2326 delete fastTrack->mBufferProvider; 2327 sq->end(false /*didModify*/); 2328 delete mFastMixer; 2329#ifdef SOAKER 2330 if (mSoaker != NULL) { 2331 mSoaker->requestExitAndWait(); 2332 } 2333 delete mSoaker; 2334#endif 2335 } 2336 delete mAudioMixer; 2337} 2338 2339class CpuStats { 2340public: 2341 CpuStats(); 2342 void sample(const String8 &title); 2343#ifdef DEBUG_CPU_USAGE 2344private: 2345 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2346 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2347 2348 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2349 2350 int mCpuNum; // thread's current CPU number 2351 int mCpukHz; // frequency of thread's current CPU in kHz 2352#endif 2353}; 2354 2355CpuStats::CpuStats() 2356#ifdef DEBUG_CPU_USAGE 2357 : mCpuNum(-1), mCpukHz(-1) 2358#endif 2359{ 2360} 2361 2362void CpuStats::sample(const String8 &title) { 2363#ifdef DEBUG_CPU_USAGE 2364 // get current thread's delta CPU time in wall clock ns 2365 double wcNs; 2366 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2367 2368 // record sample for wall clock statistics 2369 if (valid) { 2370 mWcStats.sample(wcNs); 2371 } 2372 2373 // get the current CPU number 2374 int cpuNum = sched_getcpu(); 2375 2376 // get the current CPU frequency in kHz 2377 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2378 2379 // check if either CPU number or frequency changed 2380 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2381 mCpuNum = cpuNum; 2382 mCpukHz = cpukHz; 2383 // ignore sample for purposes of cycles 2384 valid = false; 2385 } 2386 2387 // if no change in CPU number or frequency, then record sample for cycle statistics 2388 if (valid && mCpukHz > 0) { 2389 double cycles = wcNs * cpukHz * 0.000001; 2390 mHzStats.sample(cycles); 2391 } 2392 2393 unsigned n = mWcStats.n(); 2394 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2395 if ((n & 127) == 1) { 2396 long long elapsed = mCpuUsage.elapsed(); 2397 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2398 double perLoop = elapsed / (double) n; 2399 double perLoop100 = perLoop * 0.01; 2400 double perLoop1k = perLoop * 0.001; 2401 double mean = mWcStats.mean(); 2402 double stddev = mWcStats.stddev(); 2403 double minimum = mWcStats.minimum(); 2404 double maximum = mWcStats.maximum(); 2405 double meanCycles = mHzStats.mean(); 2406 double stddevCycles = mHzStats.stddev(); 2407 double minCycles = mHzStats.minimum(); 2408 double maxCycles = mHzStats.maximum(); 2409 mCpuUsage.resetElapsed(); 2410 mWcStats.reset(); 2411 mHzStats.reset(); 2412 ALOGD("CPU usage for %s over past %.1f secs\n" 2413 " (%u mixer loops at %.1f mean ms per loop):\n" 2414 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2415 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2416 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2417 title.string(), 2418 elapsed * .000000001, n, perLoop * .000001, 2419 mean * .001, 2420 stddev * .001, 2421 minimum * .001, 2422 maximum * .001, 2423 mean / perLoop100, 2424 stddev / perLoop100, 2425 minimum / perLoop100, 2426 maximum / perLoop100, 2427 meanCycles / perLoop1k, 2428 stddevCycles / perLoop1k, 2429 minCycles / perLoop1k, 2430 maxCycles / perLoop1k); 2431 2432 } 2433 } 2434#endif 2435}; 2436 2437void AudioFlinger::PlaybackThread::checkSilentMode_l() 2438{ 2439 if (!mMasterMute) { 2440 char value[PROPERTY_VALUE_MAX]; 2441 if (property_get("ro.audio.silent", value, "0") > 0) { 2442 char *endptr; 2443 unsigned long ul = strtoul(value, &endptr, 0); 2444 if (*endptr == '\0' && ul != 0) { 2445 ALOGD("Silence is golden"); 2446 // The setprop command will not allow a property to be changed after 2447 // the first time it is set, so we don't have to worry about un-muting. 2448 setMasterMute_l(true); 2449 } 2450 } 2451 } 2452} 2453 2454bool AudioFlinger::PlaybackThread::threadLoop() 2455{ 2456 Vector< sp<Track> > tracksToRemove; 2457 2458 standbyTime = systemTime(); 2459 2460 // MIXER 2461 nsecs_t lastWarning = 0; 2462if (mType == MIXER) { 2463 longStandbyExit = false; 2464} 2465 2466 // DUPLICATING 2467 // FIXME could this be made local to while loop? 2468 writeFrames = 0; 2469 2470 cacheParameters_l(); 2471 sleepTime = idleSleepTime; 2472 2473if (mType == MIXER) { 2474 sleepTimeShift = 0; 2475} 2476 2477 CpuStats cpuStats; 2478 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2479 2480 acquireWakeLock(); 2481 2482 while (!exitPending()) 2483 { 2484 cpuStats.sample(myName); 2485 2486 Vector< sp<EffectChain> > effectChains; 2487 2488 processConfigEvents(); 2489 2490 { // scope for mLock 2491 2492 Mutex::Autolock _l(mLock); 2493 2494 if (checkForNewParameters_l()) { 2495 cacheParameters_l(); 2496 } 2497 2498 saveOutputTracks(); 2499 2500 // put audio hardware into standby after short delay 2501 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2502 mSuspended > 0)) { 2503 if (!mStandby) { 2504 2505 threadLoop_standby(); 2506 2507 mStandby = true; 2508 mBytesWritten = 0; 2509 } 2510 2511 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2512 // we're about to wait, flush the binder command buffer 2513 IPCThreadState::self()->flushCommands(); 2514 2515 clearOutputTracks(); 2516 2517 if (exitPending()) break; 2518 2519 releaseWakeLock_l(); 2520 // wait until we have something to do... 2521 ALOGV("%s going to sleep", myName.string()); 2522 mWaitWorkCV.wait(mLock); 2523 ALOGV("%s waking up", myName.string()); 2524 acquireWakeLock_l(); 2525 2526 mMixerStatus = MIXER_IDLE; 2527 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2528 2529 checkSilentMode_l(); 2530 2531 standbyTime = systemTime() + standbyDelay; 2532 sleepTime = idleSleepTime; 2533 if (mType == MIXER) { 2534 sleepTimeShift = 0; 2535 } 2536 2537 continue; 2538 } 2539 } 2540 2541 // mMixerStatusIgnoringFastTracks is also updated internally 2542 mMixerStatus = prepareTracks_l(&tracksToRemove); 2543 2544 // prevent any changes in effect chain list and in each effect chain 2545 // during mixing and effect process as the audio buffers could be deleted 2546 // or modified if an effect is created or deleted 2547 lockEffectChains_l(effectChains); 2548 } 2549 2550 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2551 threadLoop_mix(); 2552 } else { 2553 threadLoop_sleepTime(); 2554 } 2555 2556 if (mSuspended > 0) { 2557 sleepTime = suspendSleepTimeUs(); 2558 } 2559 2560 // only process effects if we're going to write 2561 if (sleepTime == 0) { 2562 for (size_t i = 0; i < effectChains.size(); i ++) { 2563 effectChains[i]->process_l(); 2564 } 2565 } 2566 2567 // enable changes in effect chain 2568 unlockEffectChains(effectChains); 2569 2570 // sleepTime == 0 means we must write to audio hardware 2571 if (sleepTime == 0) { 2572 2573 threadLoop_write(); 2574 2575if (mType == MIXER) { 2576 // write blocked detection 2577 nsecs_t now = systemTime(); 2578 nsecs_t delta = now - mLastWriteTime; 2579 if (!mStandby && delta > maxPeriod) { 2580 mNumDelayedWrites++; 2581 if ((now - lastWarning) > kWarningThrottleNs) { 2582#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2583 ScopedTrace st(ATRACE_TAG, "underrun"); 2584#endif 2585 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2586 ns2ms(delta), mNumDelayedWrites, this); 2587 lastWarning = now; 2588 } 2589 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2590 // a different threshold. Or completely removed for what it is worth anyway... 2591 if (mStandby) { 2592 longStandbyExit = true; 2593 } 2594 } 2595} 2596 2597 mStandby = false; 2598 } else { 2599 usleep(sleepTime); 2600 } 2601 2602 // Finally let go of removed track(s), without the lock held 2603 // since we can't guarantee the destructors won't acquire that 2604 // same lock. This will also mutate and push a new fast mixer state. 2605 threadLoop_removeTracks(tracksToRemove); 2606 tracksToRemove.clear(); 2607 2608 // FIXME I don't understand the need for this here; 2609 // it was in the original code but maybe the 2610 // assignment in saveOutputTracks() makes this unnecessary? 2611 clearOutputTracks(); 2612 2613 // Effect chains will be actually deleted here if they were removed from 2614 // mEffectChains list during mixing or effects processing 2615 effectChains.clear(); 2616 2617 // FIXME Note that the above .clear() is no longer necessary since effectChains 2618 // is now local to this block, but will keep it for now (at least until merge done). 2619 } 2620 2621if (mType == MIXER || mType == DIRECT) { 2622 // put output stream into standby mode 2623 if (!mStandby) { 2624 mOutput->stream->common.standby(&mOutput->stream->common); 2625 } 2626} 2627if (mType == DUPLICATING) { 2628 // for DuplicatingThread, standby mode is handled by the outputTracks 2629} 2630 2631 releaseWakeLock(); 2632 2633 ALOGV("Thread %p type %d exiting", this, mType); 2634 return false; 2635} 2636 2637void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2638{ 2639 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2640} 2641 2642void AudioFlinger::MixerThread::threadLoop_write() 2643{ 2644 // FIXME we should only do one push per cycle; confirm this is true 2645 // Start the fast mixer if it's not already running 2646 if (mFastMixer != NULL) { 2647 FastMixerStateQueue *sq = mFastMixer->sq(); 2648 FastMixerState *state = sq->begin(); 2649 if (state->mCommand != FastMixerState::MIX_WRITE && 2650 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2651 if (state->mCommand == FastMixerState::COLD_IDLE) { 2652 int32_t old = android_atomic_inc(&mFastMixerFutex); 2653 if (old == -1) { 2654 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2655 } 2656 } 2657 state->mCommand = FastMixerState::MIX_WRITE; 2658 sq->end(); 2659 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2660 if (kUseFastMixer == FastMixer_Dynamic) { 2661 mNormalSink = mPipeSink; 2662 } 2663 } else { 2664 sq->end(false /*didModify*/); 2665 } 2666 } 2667 PlaybackThread::threadLoop_write(); 2668} 2669 2670// shared by MIXER and DIRECT, overridden by DUPLICATING 2671void AudioFlinger::PlaybackThread::threadLoop_write() 2672{ 2673 // FIXME rewrite to reduce number of system calls 2674 mLastWriteTime = systemTime(); 2675 mInWrite = true; 2676 int bytesWritten; 2677 2678 // If an NBAIO sink is present, use it to write the normal mixer's submix 2679 if (mNormalSink != 0) { 2680#define mBitShift 2 // FIXME 2681 size_t count = mixBufferSize >> mBitShift; 2682#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2683 Tracer::traceBegin(ATRACE_TAG, "write"); 2684#endif 2685 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2686#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2687 Tracer::traceEnd(ATRACE_TAG); 2688#endif 2689 if (framesWritten > 0) { 2690 bytesWritten = framesWritten << mBitShift; 2691 } else { 2692 bytesWritten = framesWritten; 2693 } 2694 // otherwise use the HAL / AudioStreamOut directly 2695 } else { 2696 // Direct output thread. 2697 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2698 } 2699 2700 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2701 mNumWrites++; 2702 mInWrite = false; 2703} 2704 2705void AudioFlinger::MixerThread::threadLoop_standby() 2706{ 2707 // Idle the fast mixer if it's currently running 2708 if (mFastMixer != NULL) { 2709 FastMixerStateQueue *sq = mFastMixer->sq(); 2710 FastMixerState *state = sq->begin(); 2711 if (!(state->mCommand & FastMixerState::IDLE)) { 2712 state->mCommand = FastMixerState::COLD_IDLE; 2713 state->mColdFutexAddr = &mFastMixerFutex; 2714 state->mColdGen++; 2715 mFastMixerFutex = 0; 2716 sq->end(); 2717 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2718 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2719 if (kUseFastMixer == FastMixer_Dynamic) { 2720 mNormalSink = mOutputSink; 2721 } 2722 } else { 2723 sq->end(false /*didModify*/); 2724 } 2725 } 2726 PlaybackThread::threadLoop_standby(); 2727} 2728 2729// shared by MIXER and DIRECT, overridden by DUPLICATING 2730void AudioFlinger::PlaybackThread::threadLoop_standby() 2731{ 2732 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2733 mOutput->stream->common.standby(&mOutput->stream->common); 2734} 2735 2736void AudioFlinger::MixerThread::threadLoop_mix() 2737{ 2738 // obtain the presentation timestamp of the next output buffer 2739 int64_t pts; 2740 status_t status = INVALID_OPERATION; 2741 2742 if (NULL != mOutput->stream->get_next_write_timestamp) { 2743 status = mOutput->stream->get_next_write_timestamp( 2744 mOutput->stream, &pts); 2745 } 2746 2747 if (status != NO_ERROR) { 2748 pts = AudioBufferProvider::kInvalidPTS; 2749 } 2750 2751 // mix buffers... 2752 mAudioMixer->process(pts); 2753 // increase sleep time progressively when application underrun condition clears. 2754 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2755 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2756 // such that we would underrun the audio HAL. 2757 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2758 sleepTimeShift--; 2759 } 2760 sleepTime = 0; 2761 standbyTime = systemTime() + standbyDelay; 2762 //TODO: delay standby when effects have a tail 2763} 2764 2765void AudioFlinger::MixerThread::threadLoop_sleepTime() 2766{ 2767 // If no tracks are ready, sleep once for the duration of an output 2768 // buffer size, then write 0s to the output 2769 if (sleepTime == 0) { 2770 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2771 sleepTime = activeSleepTime >> sleepTimeShift; 2772 if (sleepTime < kMinThreadSleepTimeUs) { 2773 sleepTime = kMinThreadSleepTimeUs; 2774 } 2775 // reduce sleep time in case of consecutive application underruns to avoid 2776 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2777 // duration we would end up writing less data than needed by the audio HAL if 2778 // the condition persists. 2779 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2780 sleepTimeShift++; 2781 } 2782 } else { 2783 sleepTime = idleSleepTime; 2784 } 2785 } else if (mBytesWritten != 0 || 2786 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2787 memset (mMixBuffer, 0, mixBufferSize); 2788 sleepTime = 0; 2789 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2790 } 2791 // TODO add standby time extension fct of effect tail 2792} 2793 2794// prepareTracks_l() must be called with ThreadBase::mLock held 2795AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2796 Vector< sp<Track> > *tracksToRemove) 2797{ 2798 2799 mixer_state mixerStatus = MIXER_IDLE; 2800 // find out which tracks need to be processed 2801 size_t count = mActiveTracks.size(); 2802 size_t mixedTracks = 0; 2803 size_t tracksWithEffect = 0; 2804 // counts only _active_ fast tracks 2805 size_t fastTracks = 0; 2806 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2807 2808 float masterVolume = mMasterVolume; 2809 bool masterMute = mMasterMute; 2810 2811 if (masterMute) { 2812 masterVolume = 0; 2813 } 2814 // Delegate master volume control to effect in output mix effect chain if needed 2815 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2816 if (chain != 0) { 2817 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2818 chain->setVolume_l(&v, &v); 2819 masterVolume = (float)((v + (1 << 23)) >> 24); 2820 chain.clear(); 2821 } 2822 2823 // prepare a new state to push 2824 FastMixerStateQueue *sq = NULL; 2825 FastMixerState *state = NULL; 2826 bool didModify = false; 2827 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2828 if (mFastMixer != NULL) { 2829 sq = mFastMixer->sq(); 2830 state = sq->begin(); 2831 } 2832 2833 for (size_t i=0 ; i<count ; i++) { 2834 sp<Track> t = mActiveTracks[i].promote(); 2835 if (t == 0) continue; 2836 2837 // this const just means the local variable doesn't change 2838 Track* const track = t.get(); 2839 2840 // process fast tracks 2841 if (track->isFastTrack()) { 2842 2843 // It's theoretically possible (though unlikely) for a fast track to be created 2844 // and then removed within the same normal mix cycle. This is not a problem, as 2845 // the track never becomes active so it's fast mixer slot is never touched. 2846 // The converse, of removing an (active) track and then creating a new track 2847 // at the identical fast mixer slot within the same normal mix cycle, 2848 // is impossible because the slot isn't marked available until the end of each cycle. 2849 int j = track->mFastIndex; 2850 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2851 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2852 FastTrack *fastTrack = &state->mFastTracks[j]; 2853 2854 // Determine whether the track is currently in underrun condition, 2855 // and whether it had a recent underrun. 2856 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2857 FastTrackUnderruns underruns = ftDump->mUnderruns; 2858 uint32_t recentFull = (underruns.mBitFields.mFull - 2859 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2860 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2861 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2862 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2863 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2864 uint32_t recentUnderruns = recentPartial + recentEmpty; 2865 track->mObservedUnderruns = underruns; 2866 // don't count underruns that occur while stopping or pausing 2867 // or stopped which can occur when flush() is called while active 2868 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2869 track->mUnderrunCount += recentUnderruns; 2870 } 2871 2872 // This is similar to the state machine for normal tracks, 2873 // with a few modifications for fast tracks. 2874 bool isActive = true; 2875 switch (track->mState) { 2876 case TrackBase::STOPPING_1: 2877 // track stays active in STOPPING_1 state until first underrun 2878 if (recentUnderruns > 0) { 2879 track->mState = TrackBase::STOPPING_2; 2880 } 2881 break; 2882 case TrackBase::PAUSING: 2883 // ramp down is not yet implemented 2884 track->setPaused(); 2885 break; 2886 case TrackBase::RESUMING: 2887 // ramp up is not yet implemented 2888 track->mState = TrackBase::ACTIVE; 2889 break; 2890 case TrackBase::ACTIVE: 2891 if (recentFull > 0 || recentPartial > 0) { 2892 // track has provided at least some frames recently: reset retry count 2893 track->mRetryCount = kMaxTrackRetries; 2894 } 2895 if (recentUnderruns == 0) { 2896 // no recent underruns: stay active 2897 break; 2898 } 2899 // there has recently been an underrun of some kind 2900 if (track->sharedBuffer() == 0) { 2901 // were any of the recent underruns "empty" (no frames available)? 2902 if (recentEmpty == 0) { 2903 // no, then ignore the partial underruns as they are allowed indefinitely 2904 break; 2905 } 2906 // there has recently been an "empty" underrun: decrement the retry counter 2907 if (--(track->mRetryCount) > 0) { 2908 break; 2909 } 2910 // indicate to client process that the track was disabled because of underrun; 2911 // it will then automatically call start() when data is available 2912 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2913 // remove from active list, but state remains ACTIVE [confusing but true] 2914 isActive = false; 2915 break; 2916 } 2917 // fall through 2918 case TrackBase::STOPPING_2: 2919 case TrackBase::PAUSED: 2920 case TrackBase::TERMINATED: 2921 case TrackBase::STOPPED: 2922 case TrackBase::FLUSHED: // flush() while active 2923 // Check for presentation complete if track is inactive 2924 // We have consumed all the buffers of this track. 2925 // This would be incomplete if we auto-paused on underrun 2926 { 2927 size_t audioHALFrames = 2928 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2929 size_t framesWritten = 2930 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2931 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2932 // track stays in active list until presentation is complete 2933 break; 2934 } 2935 } 2936 if (track->isStopping_2()) { 2937 track->mState = TrackBase::STOPPED; 2938 } 2939 if (track->isStopped()) { 2940 // Can't reset directly, as fast mixer is still polling this track 2941 // track->reset(); 2942 // So instead mark this track as needing to be reset after push with ack 2943 resetMask |= 1 << i; 2944 } 2945 isActive = false; 2946 break; 2947 case TrackBase::IDLE: 2948 default: 2949 LOG_FATAL("unexpected track state %d", track->mState); 2950 } 2951 2952 if (isActive) { 2953 // was it previously inactive? 2954 if (!(state->mTrackMask & (1 << j))) { 2955 ExtendedAudioBufferProvider *eabp = track; 2956 VolumeProvider *vp = track; 2957 fastTrack->mBufferProvider = eabp; 2958 fastTrack->mVolumeProvider = vp; 2959 fastTrack->mSampleRate = track->mSampleRate; 2960 fastTrack->mChannelMask = track->mChannelMask; 2961 fastTrack->mGeneration++; 2962 state->mTrackMask |= 1 << j; 2963 didModify = true; 2964 // no acknowledgement required for newly active tracks 2965 } 2966 // cache the combined master volume and stream type volume for fast mixer; this 2967 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2968 track->mCachedVolume = track->isMuted() ? 2969 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2970 ++fastTracks; 2971 } else { 2972 // was it previously active? 2973 if (state->mTrackMask & (1 << j)) { 2974 fastTrack->mBufferProvider = NULL; 2975 fastTrack->mGeneration++; 2976 state->mTrackMask &= ~(1 << j); 2977 didModify = true; 2978 // If any fast tracks were removed, we must wait for acknowledgement 2979 // because we're about to decrement the last sp<> on those tracks. 2980 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2981 } else { 2982 LOG_FATAL("fast track %d should have been active", j); 2983 } 2984 tracksToRemove->add(track); 2985 // Avoids a misleading display in dumpsys 2986 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2987 } 2988 continue; 2989 } 2990 2991 { // local variable scope to avoid goto warning 2992 2993 audio_track_cblk_t* cblk = track->cblk(); 2994 2995 // The first time a track is added we wait 2996 // for all its buffers to be filled before processing it 2997 int name = track->name(); 2998 // make sure that we have enough frames to mix one full buffer. 2999 // enforce this condition only once to enable draining the buffer in case the client 3000 // app does not call stop() and relies on underrun to stop: 3001 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3002 // during last round 3003 uint32_t minFrames = 1; 3004 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3005 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3006 if (t->sampleRate() == (int)mSampleRate) { 3007 minFrames = mNormalFrameCount; 3008 } else { 3009 // +1 for rounding and +1 for additional sample needed for interpolation 3010 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3011 // add frames already consumed but not yet released by the resampler 3012 // because cblk->framesReady() will include these frames 3013 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3014 // the minimum track buffer size is normally twice the number of frames necessary 3015 // to fill one buffer and the resampler should not leave more than one buffer worth 3016 // of unreleased frames after each pass, but just in case... 3017 ALOG_ASSERT(minFrames <= cblk->frameCount); 3018 } 3019 } 3020 if ((track->framesReady() >= minFrames) && track->isReady() && 3021 !track->isPaused() && !track->isTerminated()) 3022 { 3023 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3024 3025 mixedTracks++; 3026 3027 // track->mainBuffer() != mMixBuffer means there is an effect chain 3028 // connected to the track 3029 chain.clear(); 3030 if (track->mainBuffer() != mMixBuffer) { 3031 chain = getEffectChain_l(track->sessionId()); 3032 // Delegate volume control to effect in track effect chain if needed 3033 if (chain != 0) { 3034 tracksWithEffect++; 3035 } else { 3036 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3037 name, track->sessionId()); 3038 } 3039 } 3040 3041 3042 int param = AudioMixer::VOLUME; 3043 if (track->mFillingUpStatus == Track::FS_FILLED) { 3044 // no ramp for the first volume setting 3045 track->mFillingUpStatus = Track::FS_ACTIVE; 3046 if (track->mState == TrackBase::RESUMING) { 3047 track->mState = TrackBase::ACTIVE; 3048 param = AudioMixer::RAMP_VOLUME; 3049 } 3050 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3051 } else if (cblk->server != 0) { 3052 // If the track is stopped before the first frame was mixed, 3053 // do not apply ramp 3054 param = AudioMixer::RAMP_VOLUME; 3055 } 3056 3057 // compute volume for this track 3058 uint32_t vl, vr, va; 3059 if (track->isMuted() || track->isPausing() || 3060 mStreamTypes[track->streamType()].mute) { 3061 vl = vr = va = 0; 3062 if (track->isPausing()) { 3063 track->setPaused(); 3064 } 3065 } else { 3066 3067 // read original volumes with volume control 3068 float typeVolume = mStreamTypes[track->streamType()].volume; 3069 float v = masterVolume * typeVolume; 3070 uint32_t vlr = cblk->getVolumeLR(); 3071 vl = vlr & 0xFFFF; 3072 vr = vlr >> 16; 3073 // track volumes come from shared memory, so can't be trusted and must be clamped 3074 if (vl > MAX_GAIN_INT) { 3075 ALOGV("Track left volume out of range: %04X", vl); 3076 vl = MAX_GAIN_INT; 3077 } 3078 if (vr > MAX_GAIN_INT) { 3079 ALOGV("Track right volume out of range: %04X", vr); 3080 vr = MAX_GAIN_INT; 3081 } 3082 // now apply the master volume and stream type volume 3083 vl = (uint32_t)(v * vl) << 12; 3084 vr = (uint32_t)(v * vr) << 12; 3085 // assuming master volume and stream type volume each go up to 1.0, 3086 // vl and vr are now in 8.24 format 3087 3088 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3089 // send level comes from shared memory and so may be corrupt 3090 if (sendLevel > MAX_GAIN_INT) { 3091 ALOGV("Track send level out of range: %04X", sendLevel); 3092 sendLevel = MAX_GAIN_INT; 3093 } 3094 va = (uint32_t)(v * sendLevel); 3095 } 3096 // Delegate volume control to effect in track effect chain if needed 3097 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3098 // Do not ramp volume if volume is controlled by effect 3099 param = AudioMixer::VOLUME; 3100 track->mHasVolumeController = true; 3101 } else { 3102 // force no volume ramp when volume controller was just disabled or removed 3103 // from effect chain to avoid volume spike 3104 if (track->mHasVolumeController) { 3105 param = AudioMixer::VOLUME; 3106 } 3107 track->mHasVolumeController = false; 3108 } 3109 3110 // Convert volumes from 8.24 to 4.12 format 3111 // This additional clamping is needed in case chain->setVolume_l() overshot 3112 vl = (vl + (1 << 11)) >> 12; 3113 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3114 vr = (vr + (1 << 11)) >> 12; 3115 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3116 3117 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3118 3119 // XXX: these things DON'T need to be done each time 3120 mAudioMixer->setBufferProvider(name, track); 3121 mAudioMixer->enable(name); 3122 3123 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3124 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3125 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3126 mAudioMixer->setParameter( 3127 name, 3128 AudioMixer::TRACK, 3129 AudioMixer::FORMAT, (void *)track->format()); 3130 mAudioMixer->setParameter( 3131 name, 3132 AudioMixer::TRACK, 3133 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3134 mAudioMixer->setParameter( 3135 name, 3136 AudioMixer::RESAMPLE, 3137 AudioMixer::SAMPLE_RATE, 3138 (void *)(cblk->sampleRate)); 3139 mAudioMixer->setParameter( 3140 name, 3141 AudioMixer::TRACK, 3142 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3143 mAudioMixer->setParameter( 3144 name, 3145 AudioMixer::TRACK, 3146 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3147 3148 // reset retry count 3149 track->mRetryCount = kMaxTrackRetries; 3150 3151 // If one track is ready, set the mixer ready if: 3152 // - the mixer was not ready during previous round OR 3153 // - no other track is not ready 3154 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3155 mixerStatus != MIXER_TRACKS_ENABLED) { 3156 mixerStatus = MIXER_TRACKS_READY; 3157 } 3158 } else { 3159 // clear effect chain input buffer if an active track underruns to avoid sending 3160 // previous audio buffer again to effects 3161 chain = getEffectChain_l(track->sessionId()); 3162 if (chain != 0) { 3163 chain->clearInputBuffer(); 3164 } 3165 3166 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3167 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3168 track->isStopped() || track->isPaused()) { 3169 // We have consumed all the buffers of this track. 3170 // Remove it from the list of active tracks. 3171 // TODO: use actual buffer filling status instead of latency when available from 3172 // audio HAL 3173 size_t audioHALFrames = 3174 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3175 size_t framesWritten = 3176 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3177 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3178 if (track->isStopped()) { 3179 track->reset(); 3180 } 3181 tracksToRemove->add(track); 3182 } 3183 } else { 3184 track->mUnderrunCount++; 3185 // No buffers for this track. Give it a few chances to 3186 // fill a buffer, then remove it from active list. 3187 if (--(track->mRetryCount) <= 0) { 3188 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3189 tracksToRemove->add(track); 3190 // indicate to client process that the track was disabled because of underrun; 3191 // it will then automatically call start() when data is available 3192 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3193 // If one track is not ready, mark the mixer also not ready if: 3194 // - the mixer was ready during previous round OR 3195 // - no other track is ready 3196 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3197 mixerStatus != MIXER_TRACKS_READY) { 3198 mixerStatus = MIXER_TRACKS_ENABLED; 3199 } 3200 } 3201 mAudioMixer->disable(name); 3202 } 3203 3204 } // local variable scope to avoid goto warning 3205track_is_ready: ; 3206 3207 } 3208 3209 // Push the new FastMixer state if necessary 3210 if (didModify) { 3211 state->mFastTracksGen++; 3212 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3213 if (kUseFastMixer == FastMixer_Dynamic && 3214 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3215 state->mCommand = FastMixerState::COLD_IDLE; 3216 state->mColdFutexAddr = &mFastMixerFutex; 3217 state->mColdGen++; 3218 mFastMixerFutex = 0; 3219 if (kUseFastMixer == FastMixer_Dynamic) { 3220 mNormalSink = mOutputSink; 3221 } 3222 // If we go into cold idle, need to wait for acknowledgement 3223 // so that fast mixer stops doing I/O. 3224 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3225 } 3226 sq->end(); 3227 } 3228 if (sq != NULL) { 3229 sq->end(didModify); 3230 sq->push(block); 3231 } 3232 3233 // Now perform the deferred reset on fast tracks that have stopped 3234 while (resetMask != 0) { 3235 size_t i = __builtin_ctz(resetMask); 3236 ALOG_ASSERT(i < count); 3237 resetMask &= ~(1 << i); 3238 sp<Track> t = mActiveTracks[i].promote(); 3239 if (t == 0) continue; 3240 Track* track = t.get(); 3241 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3242 track->reset(); 3243 } 3244 3245 // remove all the tracks that need to be... 3246 count = tracksToRemove->size(); 3247 if (CC_UNLIKELY(count)) { 3248 for (size_t i=0 ; i<count ; i++) { 3249 const sp<Track>& track = tracksToRemove->itemAt(i); 3250 mActiveTracks.remove(track); 3251 if (track->mainBuffer() != mMixBuffer) { 3252 chain = getEffectChain_l(track->sessionId()); 3253 if (chain != 0) { 3254 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3255 chain->decActiveTrackCnt(); 3256 } 3257 } 3258 if (track->isTerminated()) { 3259 removeTrack_l(track); 3260 } 3261 } 3262 } 3263 3264 // mix buffer must be cleared if all tracks are connected to an 3265 // effect chain as in this case the mixer will not write to 3266 // mix buffer and track effects will accumulate into it 3267 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3268 // FIXME as a performance optimization, should remember previous zero status 3269 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3270 } 3271 3272 // if any fast tracks, then status is ready 3273 mMixerStatusIgnoringFastTracks = mixerStatus; 3274 if (fastTracks > 0) { 3275 mixerStatus = MIXER_TRACKS_READY; 3276 } 3277 return mixerStatus; 3278} 3279 3280/* 3281The derived values that are cached: 3282 - mixBufferSize from frame count * frame size 3283 - activeSleepTime from activeSleepTimeUs() 3284 - idleSleepTime from idleSleepTimeUs() 3285 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3286 - maxPeriod from frame count and sample rate (MIXER only) 3287 3288The parameters that affect these derived values are: 3289 - frame count 3290 - frame size 3291 - sample rate 3292 - device type: A2DP or not 3293 - device latency 3294 - format: PCM or not 3295 - active sleep time 3296 - idle sleep time 3297*/ 3298 3299void AudioFlinger::PlaybackThread::cacheParameters_l() 3300{ 3301 mixBufferSize = mNormalFrameCount * mFrameSize; 3302 activeSleepTime = activeSleepTimeUs(); 3303 idleSleepTime = idleSleepTimeUs(); 3304} 3305 3306void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3307{ 3308 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3309 this, streamType, mTracks.size()); 3310 Mutex::Autolock _l(mLock); 3311 3312 size_t size = mTracks.size(); 3313 for (size_t i = 0; i < size; i++) { 3314 sp<Track> t = mTracks[i]; 3315 if (t->streamType() == streamType) { 3316 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3317 t->mCblk->cv.signal(); 3318 } 3319 } 3320} 3321 3322// getTrackName_l() must be called with ThreadBase::mLock held 3323int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3324{ 3325 return mAudioMixer->getTrackName(channelMask); 3326} 3327 3328// deleteTrackName_l() must be called with ThreadBase::mLock held 3329void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3330{ 3331 ALOGV("remove track (%d) and delete from mixer", name); 3332 mAudioMixer->deleteTrackName(name); 3333} 3334 3335// checkForNewParameters_l() must be called with ThreadBase::mLock held 3336bool AudioFlinger::MixerThread::checkForNewParameters_l() 3337{ 3338 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3339 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3340 bool reconfig = false; 3341 3342 while (!mNewParameters.isEmpty()) { 3343 3344 if (mFastMixer != NULL) { 3345 FastMixerStateQueue *sq = mFastMixer->sq(); 3346 FastMixerState *state = sq->begin(); 3347 if (!(state->mCommand & FastMixerState::IDLE)) { 3348 previousCommand = state->mCommand; 3349 state->mCommand = FastMixerState::HOT_IDLE; 3350 sq->end(); 3351 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3352 } else { 3353 sq->end(false /*didModify*/); 3354 } 3355 } 3356 3357 status_t status = NO_ERROR; 3358 String8 keyValuePair = mNewParameters[0]; 3359 AudioParameter param = AudioParameter(keyValuePair); 3360 int value; 3361 3362 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3363 reconfig = true; 3364 } 3365 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3366 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3367 status = BAD_VALUE; 3368 } else { 3369 reconfig = true; 3370 } 3371 } 3372 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3373 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3374 status = BAD_VALUE; 3375 } else { 3376 reconfig = true; 3377 } 3378 } 3379 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3380 // do not accept frame count changes if tracks are open as the track buffer 3381 // size depends on frame count and correct behavior would not be guaranteed 3382 // if frame count is changed after track creation 3383 if (!mTracks.isEmpty()) { 3384 status = INVALID_OPERATION; 3385 } else { 3386 reconfig = true; 3387 } 3388 } 3389 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3390#ifdef ADD_BATTERY_DATA 3391 // when changing the audio output device, call addBatteryData to notify 3392 // the change 3393 if ((int)mDevice != value) { 3394 uint32_t params = 0; 3395 // check whether speaker is on 3396 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3397 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3398 } 3399 3400 int deviceWithoutSpeaker 3401 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3402 // check if any other device (except speaker) is on 3403 if (value & deviceWithoutSpeaker ) { 3404 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3405 } 3406 3407 if (params != 0) { 3408 addBatteryData(params); 3409 } 3410 } 3411#endif 3412 3413 // forward device change to effects that have requested to be 3414 // aware of attached audio device. 3415 mDevice = (uint32_t)value; 3416 for (size_t i = 0; i < mEffectChains.size(); i++) { 3417 mEffectChains[i]->setDevice_l(mDevice); 3418 } 3419 } 3420 3421 if (status == NO_ERROR) { 3422 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3423 keyValuePair.string()); 3424 if (!mStandby && status == INVALID_OPERATION) { 3425 mOutput->stream->common.standby(&mOutput->stream->common); 3426 mStandby = true; 3427 mBytesWritten = 0; 3428 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3429 keyValuePair.string()); 3430 } 3431 if (status == NO_ERROR && reconfig) { 3432 delete mAudioMixer; 3433 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3434 mAudioMixer = NULL; 3435 readOutputParameters(); 3436 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3437 for (size_t i = 0; i < mTracks.size() ; i++) { 3438 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3439 if (name < 0) break; 3440 mTracks[i]->mName = name; 3441 // limit track sample rate to 2 x new output sample rate 3442 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3443 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3444 } 3445 } 3446 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3447 } 3448 } 3449 3450 mNewParameters.removeAt(0); 3451 3452 mParamStatus = status; 3453 mParamCond.signal(); 3454 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3455 // already timed out waiting for the status and will never signal the condition. 3456 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3457 } 3458 3459 if (!(previousCommand & FastMixerState::IDLE)) { 3460 ALOG_ASSERT(mFastMixer != NULL); 3461 FastMixerStateQueue *sq = mFastMixer->sq(); 3462 FastMixerState *state = sq->begin(); 3463 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3464 state->mCommand = previousCommand; 3465 sq->end(); 3466 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3467 } 3468 3469 return reconfig; 3470} 3471 3472status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3473{ 3474 const size_t SIZE = 256; 3475 char buffer[SIZE]; 3476 String8 result; 3477 3478 PlaybackThread::dumpInternals(fd, args); 3479 3480 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3481 result.append(buffer); 3482 write(fd, result.string(), result.size()); 3483 3484 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3485 FastMixerDumpState copy = mFastMixerDumpState; 3486 copy.dump(fd); 3487 3488#ifdef STATE_QUEUE_DUMP 3489 // Similar for state queue 3490 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3491 observerCopy.dump(fd); 3492 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3493 mutatorCopy.dump(fd); 3494#endif 3495 3496 // Write the tee output to a .wav file 3497 NBAIO_Source *teeSource = mTeeSource.get(); 3498 if (teeSource != NULL) { 3499 char teePath[64]; 3500 struct timeval tv; 3501 gettimeofday(&tv, NULL); 3502 struct tm tm; 3503 localtime_r(&tv.tv_sec, &tm); 3504 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3505 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3506 if (teeFd >= 0) { 3507 char wavHeader[44]; 3508 memcpy(wavHeader, 3509 "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0", 3510 sizeof(wavHeader)); 3511 NBAIO_Format format = teeSource->format(); 3512 unsigned channelCount = Format_channelCount(format); 3513 ALOG_ASSERT(channelCount <= FCC_2); 3514 unsigned sampleRate = Format_sampleRate(format); 3515 wavHeader[22] = channelCount; // number of channels 3516 wavHeader[24] = sampleRate; // sample rate 3517 wavHeader[25] = sampleRate >> 8; 3518 wavHeader[32] = channelCount * 2; // block alignment 3519 write(teeFd, wavHeader, sizeof(wavHeader)); 3520 size_t total = 0; 3521 bool firstRead = true; 3522 for (;;) { 3523#define TEE_SINK_READ 1024 3524 short buffer[TEE_SINK_READ * FCC_2]; 3525 size_t count = TEE_SINK_READ; 3526 ssize_t actual = teeSource->read(buffer, count); 3527 bool wasFirstRead = firstRead; 3528 firstRead = false; 3529 if (actual <= 0) { 3530 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3531 continue; 3532 } 3533 break; 3534 } 3535 ALOG_ASSERT(actual <= count); 3536 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3537 total += actual; 3538 } 3539 lseek(teeFd, (off_t) 4, SEEK_SET); 3540 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3541 write(teeFd, &temp, sizeof(temp)); 3542 lseek(teeFd, (off_t) 40, SEEK_SET); 3543 temp = total * channelCount * sizeof(short); 3544 write(teeFd, &temp, sizeof(temp)); 3545 close(teeFd); 3546 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3547 } else { 3548 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3549 } 3550 } 3551 3552 return NO_ERROR; 3553} 3554 3555uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3556{ 3557 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3558} 3559 3560uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3561{ 3562 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3563} 3564 3565void AudioFlinger::MixerThread::cacheParameters_l() 3566{ 3567 PlaybackThread::cacheParameters_l(); 3568 3569 // FIXME: Relaxed timing because of a certain device that can't meet latency 3570 // Should be reduced to 2x after the vendor fixes the driver issue 3571 // increase threshold again due to low power audio mode. The way this warning 3572 // threshold is calculated and its usefulness should be reconsidered anyway. 3573 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3574} 3575 3576// ---------------------------------------------------------------------------- 3577AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3578 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3579 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3580 // mLeftVolFloat, mRightVolFloat 3581{ 3582} 3583 3584AudioFlinger::DirectOutputThread::~DirectOutputThread() 3585{ 3586} 3587 3588AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3589 Vector< sp<Track> > *tracksToRemove 3590) 3591{ 3592 sp<Track> trackToRemove; 3593 3594 mixer_state mixerStatus = MIXER_IDLE; 3595 3596 // find out which tracks need to be processed 3597 if (mActiveTracks.size() != 0) { 3598 sp<Track> t = mActiveTracks[0].promote(); 3599 // The track died recently 3600 if (t == 0) return MIXER_IDLE; 3601 3602 Track* const track = t.get(); 3603 audio_track_cblk_t* cblk = track->cblk(); 3604 3605 // The first time a track is added we wait 3606 // for all its buffers to be filled before processing it 3607 uint32_t minFrames; 3608 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3609 minFrames = mNormalFrameCount; 3610 } else { 3611 minFrames = 1; 3612 } 3613 if ((track->framesReady() >= minFrames) && track->isReady() && 3614 !track->isPaused() && !track->isTerminated()) 3615 { 3616 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3617 3618 if (track->mFillingUpStatus == Track::FS_FILLED) { 3619 track->mFillingUpStatus = Track::FS_ACTIVE; 3620 mLeftVolFloat = mRightVolFloat = 0; 3621 if (track->mState == TrackBase::RESUMING) { 3622 track->mState = TrackBase::ACTIVE; 3623 } 3624 } 3625 3626 // compute volume for this track 3627 float left, right; 3628 if (track->isMuted() || mMasterMute || track->isPausing() || 3629 mStreamTypes[track->streamType()].mute) { 3630 left = right = 0; 3631 if (track->isPausing()) { 3632 track->setPaused(); 3633 } 3634 } else { 3635 float typeVolume = mStreamTypes[track->streamType()].volume; 3636 float v = mMasterVolume * typeVolume; 3637 uint32_t vlr = cblk->getVolumeLR(); 3638 float v_clamped = v * (vlr & 0xFFFF); 3639 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3640 left = v_clamped/MAX_GAIN; 3641 v_clamped = v * (vlr >> 16); 3642 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3643 right = v_clamped/MAX_GAIN; 3644 } 3645 3646 if (left != mLeftVolFloat || right != mRightVolFloat) { 3647 mLeftVolFloat = left; 3648 mRightVolFloat = right; 3649 3650 // Convert volumes from float to 8.24 3651 uint32_t vl = (uint32_t)(left * (1 << 24)); 3652 uint32_t vr = (uint32_t)(right * (1 << 24)); 3653 3654 // Delegate volume control to effect in track effect chain if needed 3655 // only one effect chain can be present on DirectOutputThread, so if 3656 // there is one, the track is connected to it 3657 if (!mEffectChains.isEmpty()) { 3658 // Do not ramp volume if volume is controlled by effect 3659 mEffectChains[0]->setVolume_l(&vl, &vr); 3660 left = (float)vl / (1 << 24); 3661 right = (float)vr / (1 << 24); 3662 } 3663 mOutput->stream->set_volume(mOutput->stream, left, right); 3664 } 3665 3666 // reset retry count 3667 track->mRetryCount = kMaxTrackRetriesDirect; 3668 mActiveTrack = t; 3669 mixerStatus = MIXER_TRACKS_READY; 3670 } else { 3671 // clear effect chain input buffer if an active track underruns to avoid sending 3672 // previous audio buffer again to effects 3673 if (!mEffectChains.isEmpty()) { 3674 mEffectChains[0]->clearInputBuffer(); 3675 } 3676 3677 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3678 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3679 track->isStopped() || track->isPaused()) { 3680 // We have consumed all the buffers of this track. 3681 // Remove it from the list of active tracks. 3682 // TODO: implement behavior for compressed audio 3683 size_t audioHALFrames = 3684 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3685 size_t framesWritten = 3686 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3687 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3688 if (track->isStopped()) { 3689 track->reset(); 3690 } 3691 trackToRemove = track; 3692 } 3693 } else { 3694 // No buffers for this track. Give it a few chances to 3695 // fill a buffer, then remove it from active list. 3696 if (--(track->mRetryCount) <= 0) { 3697 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3698 trackToRemove = track; 3699 } else { 3700 mixerStatus = MIXER_TRACKS_ENABLED; 3701 } 3702 } 3703 } 3704 } 3705 3706 // FIXME merge this with similar code for removing multiple tracks 3707 // remove all the tracks that need to be... 3708 if (CC_UNLIKELY(trackToRemove != 0)) { 3709 tracksToRemove->add(trackToRemove); 3710 mActiveTracks.remove(trackToRemove); 3711 if (!mEffectChains.isEmpty()) { 3712 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3713 trackToRemove->sessionId()); 3714 mEffectChains[0]->decActiveTrackCnt(); 3715 } 3716 if (trackToRemove->isTerminated()) { 3717 removeTrack_l(trackToRemove); 3718 } 3719 } 3720 3721 return mixerStatus; 3722} 3723 3724void AudioFlinger::DirectOutputThread::threadLoop_mix() 3725{ 3726 AudioBufferProvider::Buffer buffer; 3727 size_t frameCount = mFrameCount; 3728 int8_t *curBuf = (int8_t *)mMixBuffer; 3729 // output audio to hardware 3730 while (frameCount) { 3731 buffer.frameCount = frameCount; 3732 mActiveTrack->getNextBuffer(&buffer); 3733 if (CC_UNLIKELY(buffer.raw == NULL)) { 3734 memset(curBuf, 0, frameCount * mFrameSize); 3735 break; 3736 } 3737 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3738 frameCount -= buffer.frameCount; 3739 curBuf += buffer.frameCount * mFrameSize; 3740 mActiveTrack->releaseBuffer(&buffer); 3741 } 3742 sleepTime = 0; 3743 standbyTime = systemTime() + standbyDelay; 3744 mActiveTrack.clear(); 3745 3746} 3747 3748void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3749{ 3750 if (sleepTime == 0) { 3751 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3752 sleepTime = activeSleepTime; 3753 } else { 3754 sleepTime = idleSleepTime; 3755 } 3756 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3757 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3758 sleepTime = 0; 3759 } 3760} 3761 3762// getTrackName_l() must be called with ThreadBase::mLock held 3763int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3764{ 3765 return 0; 3766} 3767 3768// deleteTrackName_l() must be called with ThreadBase::mLock held 3769void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3770{ 3771} 3772 3773// checkForNewParameters_l() must be called with ThreadBase::mLock held 3774bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3775{ 3776 bool reconfig = false; 3777 3778 while (!mNewParameters.isEmpty()) { 3779 status_t status = NO_ERROR; 3780 String8 keyValuePair = mNewParameters[0]; 3781 AudioParameter param = AudioParameter(keyValuePair); 3782 int value; 3783 3784 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3785 // do not accept frame count changes if tracks are open as the track buffer 3786 // size depends on frame count and correct behavior would not be garantied 3787 // if frame count is changed after track creation 3788 if (!mTracks.isEmpty()) { 3789 status = INVALID_OPERATION; 3790 } else { 3791 reconfig = true; 3792 } 3793 } 3794 if (status == NO_ERROR) { 3795 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3796 keyValuePair.string()); 3797 if (!mStandby && status == INVALID_OPERATION) { 3798 mOutput->stream->common.standby(&mOutput->stream->common); 3799 mStandby = true; 3800 mBytesWritten = 0; 3801 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3802 keyValuePair.string()); 3803 } 3804 if (status == NO_ERROR && reconfig) { 3805 readOutputParameters(); 3806 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3807 } 3808 } 3809 3810 mNewParameters.removeAt(0); 3811 3812 mParamStatus = status; 3813 mParamCond.signal(); 3814 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3815 // already timed out waiting for the status and will never signal the condition. 3816 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3817 } 3818 return reconfig; 3819} 3820 3821uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3822{ 3823 uint32_t time; 3824 if (audio_is_linear_pcm(mFormat)) { 3825 time = PlaybackThread::activeSleepTimeUs(); 3826 } else { 3827 time = 10000; 3828 } 3829 return time; 3830} 3831 3832uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3833{ 3834 uint32_t time; 3835 if (audio_is_linear_pcm(mFormat)) { 3836 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3837 } else { 3838 time = 10000; 3839 } 3840 return time; 3841} 3842 3843uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3844{ 3845 uint32_t time; 3846 if (audio_is_linear_pcm(mFormat)) { 3847 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3848 } else { 3849 time = 10000; 3850 } 3851 return time; 3852} 3853 3854void AudioFlinger::DirectOutputThread::cacheParameters_l() 3855{ 3856 PlaybackThread::cacheParameters_l(); 3857 3858 // use shorter standby delay as on normal output to release 3859 // hardware resources as soon as possible 3860 standbyDelay = microseconds(activeSleepTime*2); 3861} 3862 3863// ---------------------------------------------------------------------------- 3864 3865AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3866 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3867 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3868 mWaitTimeMs(UINT_MAX) 3869{ 3870 addOutputTrack(mainThread); 3871} 3872 3873AudioFlinger::DuplicatingThread::~DuplicatingThread() 3874{ 3875 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3876 mOutputTracks[i]->destroy(); 3877 } 3878} 3879 3880void AudioFlinger::DuplicatingThread::threadLoop_mix() 3881{ 3882 // mix buffers... 3883 if (outputsReady(outputTracks)) { 3884 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3885 } else { 3886 memset(mMixBuffer, 0, mixBufferSize); 3887 } 3888 sleepTime = 0; 3889 writeFrames = mNormalFrameCount; 3890} 3891 3892void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3893{ 3894 if (sleepTime == 0) { 3895 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3896 sleepTime = activeSleepTime; 3897 } else { 3898 sleepTime = idleSleepTime; 3899 } 3900 } else if (mBytesWritten != 0) { 3901 // flush remaining overflow buffers in output tracks 3902 for (size_t i = 0; i < outputTracks.size(); i++) { 3903 if (outputTracks[i]->isActive()) { 3904 sleepTime = 0; 3905 writeFrames = 0; 3906 memset(mMixBuffer, 0, mixBufferSize); 3907 break; 3908 } 3909 } 3910 } 3911} 3912 3913void AudioFlinger::DuplicatingThread::threadLoop_write() 3914{ 3915 standbyTime = systemTime() + standbyDelay; 3916 for (size_t i = 0; i < outputTracks.size(); i++) { 3917 outputTracks[i]->write(mMixBuffer, writeFrames); 3918 } 3919 mBytesWritten += mixBufferSize; 3920} 3921 3922void AudioFlinger::DuplicatingThread::threadLoop_standby() 3923{ 3924 // DuplicatingThread implements standby by stopping all tracks 3925 for (size_t i = 0; i < outputTracks.size(); i++) { 3926 outputTracks[i]->stop(); 3927 } 3928} 3929 3930void AudioFlinger::DuplicatingThread::saveOutputTracks() 3931{ 3932 outputTracks = mOutputTracks; 3933} 3934 3935void AudioFlinger::DuplicatingThread::clearOutputTracks() 3936{ 3937 outputTracks.clear(); 3938} 3939 3940void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3941{ 3942 Mutex::Autolock _l(mLock); 3943 // FIXME explain this formula 3944 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3945 OutputTrack *outputTrack = new OutputTrack(thread, 3946 this, 3947 mSampleRate, 3948 mFormat, 3949 mChannelMask, 3950 frameCount); 3951 if (outputTrack->cblk() != NULL) { 3952 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3953 mOutputTracks.add(outputTrack); 3954 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3955 updateWaitTime_l(); 3956 } 3957} 3958 3959void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3960{ 3961 Mutex::Autolock _l(mLock); 3962 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3963 if (mOutputTracks[i]->thread() == thread) { 3964 mOutputTracks[i]->destroy(); 3965 mOutputTracks.removeAt(i); 3966 updateWaitTime_l(); 3967 return; 3968 } 3969 } 3970 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3971} 3972 3973// caller must hold mLock 3974void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3975{ 3976 mWaitTimeMs = UINT_MAX; 3977 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3978 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3979 if (strong != 0) { 3980 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3981 if (waitTimeMs < mWaitTimeMs) { 3982 mWaitTimeMs = waitTimeMs; 3983 } 3984 } 3985 } 3986} 3987 3988 3989bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3990{ 3991 for (size_t i = 0; i < outputTracks.size(); i++) { 3992 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3993 if (thread == 0) { 3994 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3995 return false; 3996 } 3997 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3998 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3999 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4000 return false; 4001 } 4002 } 4003 return true; 4004} 4005 4006uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4007{ 4008 return (mWaitTimeMs * 1000) / 2; 4009} 4010 4011void AudioFlinger::DuplicatingThread::cacheParameters_l() 4012{ 4013 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4014 updateWaitTime_l(); 4015 4016 MixerThread::cacheParameters_l(); 4017} 4018 4019// ---------------------------------------------------------------------------- 4020 4021// TrackBase constructor must be called with AudioFlinger::mLock held 4022AudioFlinger::ThreadBase::TrackBase::TrackBase( 4023 ThreadBase *thread, 4024 const sp<Client>& client, 4025 uint32_t sampleRate, 4026 audio_format_t format, 4027 uint32_t channelMask, 4028 int frameCount, 4029 const sp<IMemory>& sharedBuffer, 4030 int sessionId) 4031 : RefBase(), 4032 mThread(thread), 4033 mClient(client), 4034 mCblk(NULL), 4035 // mBuffer 4036 // mBufferEnd 4037 mFrameCount(0), 4038 mState(IDLE), 4039 mSampleRate(sampleRate), 4040 mFormat(format), 4041 mStepServerFailed(false), 4042 mSessionId(sessionId) 4043 // mChannelCount 4044 // mChannelMask 4045{ 4046 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4047 4048 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4049 size_t size = sizeof(audio_track_cblk_t); 4050 uint8_t channelCount = popcount(channelMask); 4051 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4052 if (sharedBuffer == 0) { 4053 size += bufferSize; 4054 } 4055 4056 if (client != NULL) { 4057 mCblkMemory = client->heap()->allocate(size); 4058 if (mCblkMemory != 0) { 4059 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4060 if (mCblk != NULL) { // construct the shared structure in-place. 4061 new(mCblk) audio_track_cblk_t(); 4062 // clear all buffers 4063 mCblk->frameCount = frameCount; 4064 mCblk->sampleRate = sampleRate; 4065// uncomment the following lines to quickly test 32-bit wraparound 4066// mCblk->user = 0xffff0000; 4067// mCblk->server = 0xffff0000; 4068// mCblk->userBase = 0xffff0000; 4069// mCblk->serverBase = 0xffff0000; 4070 mChannelCount = channelCount; 4071 mChannelMask = channelMask; 4072 if (sharedBuffer == 0) { 4073 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4074 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4075 // Force underrun condition to avoid false underrun callback until first data is 4076 // written to buffer (other flags are cleared) 4077 mCblk->flags = CBLK_UNDERRUN_ON; 4078 } else { 4079 mBuffer = sharedBuffer->pointer(); 4080 } 4081 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4082 } 4083 } else { 4084 ALOGE("not enough memory for AudioTrack size=%u", size); 4085 client->heap()->dump("AudioTrack"); 4086 return; 4087 } 4088 } else { 4089 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4090 // construct the shared structure in-place. 4091 new(mCblk) audio_track_cblk_t(); 4092 // clear all buffers 4093 mCblk->frameCount = frameCount; 4094 mCblk->sampleRate = sampleRate; 4095// uncomment the following lines to quickly test 32-bit wraparound 4096// mCblk->user = 0xffff0000; 4097// mCblk->server = 0xffff0000; 4098// mCblk->userBase = 0xffff0000; 4099// mCblk->serverBase = 0xffff0000; 4100 mChannelCount = channelCount; 4101 mChannelMask = channelMask; 4102 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4103 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4104 // Force underrun condition to avoid false underrun callback until first data is 4105 // written to buffer (other flags are cleared) 4106 mCblk->flags = CBLK_UNDERRUN_ON; 4107 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4108 } 4109} 4110 4111AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4112{ 4113 if (mCblk != NULL) { 4114 if (mClient == 0) { 4115 delete mCblk; 4116 } else { 4117 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4118 } 4119 } 4120 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4121 if (mClient != 0) { 4122 // Client destructor must run with AudioFlinger mutex locked 4123 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4124 // If the client's reference count drops to zero, the associated destructor 4125 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4126 // relying on the automatic clear() at end of scope. 4127 mClient.clear(); 4128 } 4129} 4130 4131// AudioBufferProvider interface 4132// getNextBuffer() = 0; 4133// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4134void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4135{ 4136 buffer->raw = NULL; 4137 mFrameCount = buffer->frameCount; 4138 // FIXME See note at getNextBuffer() 4139 (void) step(); // ignore return value of step() 4140 buffer->frameCount = 0; 4141} 4142 4143bool AudioFlinger::ThreadBase::TrackBase::step() { 4144 bool result; 4145 audio_track_cblk_t* cblk = this->cblk(); 4146 4147 result = cblk->stepServer(mFrameCount); 4148 if (!result) { 4149 ALOGV("stepServer failed acquiring cblk mutex"); 4150 mStepServerFailed = true; 4151 } 4152 return result; 4153} 4154 4155void AudioFlinger::ThreadBase::TrackBase::reset() { 4156 audio_track_cblk_t* cblk = this->cblk(); 4157 4158 cblk->user = 0; 4159 cblk->server = 0; 4160 cblk->userBase = 0; 4161 cblk->serverBase = 0; 4162 mStepServerFailed = false; 4163 ALOGV("TrackBase::reset"); 4164} 4165 4166int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4167 return (int)mCblk->sampleRate; 4168} 4169 4170void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4171 audio_track_cblk_t* cblk = this->cblk(); 4172 size_t frameSize = cblk->frameSize; 4173 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4174 int8_t *bufferEnd = bufferStart + frames * frameSize; 4175 4176 // Check validity of returned pointer in case the track control block would have been corrupted. 4177 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4178 "TrackBase::getBuffer buffer out of range:\n" 4179 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4180 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4181 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4182 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4183 4184 return bufferStart; 4185} 4186 4187status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4188{ 4189 mSyncEvents.add(event); 4190 return NO_ERROR; 4191} 4192 4193// ---------------------------------------------------------------------------- 4194 4195// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4196AudioFlinger::PlaybackThread::Track::Track( 4197 PlaybackThread *thread, 4198 const sp<Client>& client, 4199 audio_stream_type_t streamType, 4200 uint32_t sampleRate, 4201 audio_format_t format, 4202 uint32_t channelMask, 4203 int frameCount, 4204 const sp<IMemory>& sharedBuffer, 4205 int sessionId, 4206 IAudioFlinger::track_flags_t flags) 4207 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4208 mMute(false), 4209 mFillingUpStatus(FS_INVALID), 4210 // mRetryCount initialized later when needed 4211 mSharedBuffer(sharedBuffer), 4212 mStreamType(streamType), 4213 mName(-1), // see note below 4214 mMainBuffer(thread->mixBuffer()), 4215 mAuxBuffer(NULL), 4216 mAuxEffectId(0), mHasVolumeController(false), 4217 mPresentationCompleteFrames(0), 4218 mFlags(flags), 4219 mFastIndex(-1), 4220 mUnderrunCount(0), 4221 mCachedVolume(1.0) 4222{ 4223 if (mCblk != NULL) { 4224 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4225 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4226 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4227 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4228 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4229 if (mName < 0) { 4230 ALOGE("no more track names available"); 4231 return; 4232 } 4233 // only allocate a fast track index if we were able to allocate a normal track name 4234 if (flags & IAudioFlinger::TRACK_FAST) { 4235 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4236 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4237 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4238 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4239 // FIXME This is too eager. We allocate a fast track index before the 4240 // fast track becomes active. Since fast tracks are a scarce resource, 4241 // this means we are potentially denying other more important fast tracks from 4242 // being created. It would be better to allocate the index dynamically. 4243 mFastIndex = i; 4244 // Read the initial underruns because this field is never cleared by the fast mixer 4245 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4246 thread->mFastTrackAvailMask &= ~(1 << i); 4247 } 4248 } 4249 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4250} 4251 4252AudioFlinger::PlaybackThread::Track::~Track() 4253{ 4254 ALOGV("PlaybackThread::Track destructor"); 4255 sp<ThreadBase> thread = mThread.promote(); 4256 if (thread != 0) { 4257 Mutex::Autolock _l(thread->mLock); 4258 mState = TERMINATED; 4259 } 4260} 4261 4262void AudioFlinger::PlaybackThread::Track::destroy() 4263{ 4264 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4265 // by removing it from mTracks vector, so there is a risk that this Tracks's 4266 // destructor is called. As the destructor needs to lock mLock, 4267 // we must acquire a strong reference on this Track before locking mLock 4268 // here so that the destructor is called only when exiting this function. 4269 // On the other hand, as long as Track::destroy() is only called by 4270 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4271 // this Track with its member mTrack. 4272 sp<Track> keep(this); 4273 { // scope for mLock 4274 sp<ThreadBase> thread = mThread.promote(); 4275 if (thread != 0) { 4276 if (!isOutputTrack()) { 4277 if (mState == ACTIVE || mState == RESUMING) { 4278 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4279 4280#ifdef ADD_BATTERY_DATA 4281 // to track the speaker usage 4282 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4283#endif 4284 } 4285 AudioSystem::releaseOutput(thread->id()); 4286 } 4287 Mutex::Autolock _l(thread->mLock); 4288 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4289 playbackThread->destroyTrack_l(this); 4290 } 4291 } 4292} 4293 4294/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4295{ 4296 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4297 " Server User Main buf Aux Buf Flags Underruns\n"); 4298} 4299 4300void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4301{ 4302 uint32_t vlr = mCblk->getVolumeLR(); 4303 if (isFastTrack()) { 4304 sprintf(buffer, " F %2d", mFastIndex); 4305 } else { 4306 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4307 } 4308 track_state state = mState; 4309 char stateChar; 4310 switch (state) { 4311 case IDLE: 4312 stateChar = 'I'; 4313 break; 4314 case TERMINATED: 4315 stateChar = 'T'; 4316 break; 4317 case STOPPING_1: 4318 stateChar = 's'; 4319 break; 4320 case STOPPING_2: 4321 stateChar = '5'; 4322 break; 4323 case STOPPED: 4324 stateChar = 'S'; 4325 break; 4326 case RESUMING: 4327 stateChar = 'R'; 4328 break; 4329 case ACTIVE: 4330 stateChar = 'A'; 4331 break; 4332 case PAUSING: 4333 stateChar = 'p'; 4334 break; 4335 case PAUSED: 4336 stateChar = 'P'; 4337 break; 4338 case FLUSHED: 4339 stateChar = 'F'; 4340 break; 4341 default: 4342 stateChar = '?'; 4343 break; 4344 } 4345 char nowInUnderrun; 4346 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4347 case UNDERRUN_FULL: 4348 nowInUnderrun = ' '; 4349 break; 4350 case UNDERRUN_PARTIAL: 4351 nowInUnderrun = '<'; 4352 break; 4353 case UNDERRUN_EMPTY: 4354 nowInUnderrun = '*'; 4355 break; 4356 default: 4357 nowInUnderrun = '?'; 4358 break; 4359 } 4360 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4361 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4362 (mClient == 0) ? getpid_cached : mClient->pid(), 4363 mStreamType, 4364 mFormat, 4365 mChannelMask, 4366 mSessionId, 4367 mFrameCount, 4368 mCblk->frameCount, 4369 stateChar, 4370 mMute, 4371 mFillingUpStatus, 4372 mCblk->sampleRate, 4373 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4374 20.0 * log10((vlr >> 16) / 4096.0), 4375 mCblk->server, 4376 mCblk->user, 4377 (int)mMainBuffer, 4378 (int)mAuxBuffer, 4379 mCblk->flags, 4380 mUnderrunCount, 4381 nowInUnderrun); 4382} 4383 4384// AudioBufferProvider interface 4385status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4386 AudioBufferProvider::Buffer* buffer, int64_t pts) 4387{ 4388 audio_track_cblk_t* cblk = this->cblk(); 4389 uint32_t framesReady; 4390 uint32_t framesReq = buffer->frameCount; 4391 4392 // Check if last stepServer failed, try to step now 4393 if (mStepServerFailed) { 4394 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4395 // Since the fast mixer is higher priority than client callback thread, 4396 // it does not result in priority inversion for client. 4397 // But a non-blocking solution would be preferable to avoid 4398 // fast mixer being unable to tryLock(), and 4399 // to avoid the extra context switches if the client wakes up, 4400 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4401 if (!step()) goto getNextBuffer_exit; 4402 ALOGV("stepServer recovered"); 4403 mStepServerFailed = false; 4404 } 4405 4406 // FIXME Same as above 4407 framesReady = cblk->framesReady(); 4408 4409 if (CC_LIKELY(framesReady)) { 4410 uint32_t s = cblk->server; 4411 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4412 4413 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4414 if (framesReq > framesReady) { 4415 framesReq = framesReady; 4416 } 4417 if (framesReq > bufferEnd - s) { 4418 framesReq = bufferEnd - s; 4419 } 4420 4421 buffer->raw = getBuffer(s, framesReq); 4422 if (buffer->raw == NULL) goto getNextBuffer_exit; 4423 4424 buffer->frameCount = framesReq; 4425 return NO_ERROR; 4426 } 4427 4428getNextBuffer_exit: 4429 buffer->raw = NULL; 4430 buffer->frameCount = 0; 4431 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4432 return NOT_ENOUGH_DATA; 4433} 4434 4435// Note that framesReady() takes a mutex on the control block using tryLock(). 4436// This could result in priority inversion if framesReady() is called by the normal mixer, 4437// as the normal mixer thread runs at lower 4438// priority than the client's callback thread: there is a short window within framesReady() 4439// during which the normal mixer could be preempted, and the client callback would block. 4440// Another problem can occur if framesReady() is called by the fast mixer: 4441// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4442// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4443size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4444 return mCblk->framesReady(); 4445} 4446 4447// Don't call for fast tracks; the framesReady() could result in priority inversion 4448bool AudioFlinger::PlaybackThread::Track::isReady() const { 4449 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4450 4451 if (framesReady() >= mCblk->frameCount || 4452 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4453 mFillingUpStatus = FS_FILLED; 4454 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4455 return true; 4456 } 4457 return false; 4458} 4459 4460status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4461 int triggerSession) 4462{ 4463 status_t status = NO_ERROR; 4464 ALOGV("start(%d), calling pid %d session %d", 4465 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4466 4467 sp<ThreadBase> thread = mThread.promote(); 4468 if (thread != 0) { 4469 Mutex::Autolock _l(thread->mLock); 4470 track_state state = mState; 4471 // here the track could be either new, or restarted 4472 // in both cases "unstop" the track 4473 if (mState == PAUSED) { 4474 mState = TrackBase::RESUMING; 4475 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4476 } else { 4477 mState = TrackBase::ACTIVE; 4478 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4479 } 4480 4481 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4482 thread->mLock.unlock(); 4483 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4484 thread->mLock.lock(); 4485 4486#ifdef ADD_BATTERY_DATA 4487 // to track the speaker usage 4488 if (status == NO_ERROR) { 4489 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4490 } 4491#endif 4492 } 4493 if (status == NO_ERROR) { 4494 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4495 playbackThread->addTrack_l(this); 4496 } else { 4497 mState = state; 4498 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4499 } 4500 } else { 4501 status = BAD_VALUE; 4502 } 4503 return status; 4504} 4505 4506void AudioFlinger::PlaybackThread::Track::stop() 4507{ 4508 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4509 sp<ThreadBase> thread = mThread.promote(); 4510 if (thread != 0) { 4511 Mutex::Autolock _l(thread->mLock); 4512 track_state state = mState; 4513 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4514 // If the track is not active (PAUSED and buffers full), flush buffers 4515 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4516 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4517 reset(); 4518 mState = STOPPED; 4519 } else if (!isFastTrack()) { 4520 mState = STOPPED; 4521 } else { 4522 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4523 // and then to STOPPED and reset() when presentation is complete 4524 mState = STOPPING_1; 4525 } 4526 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4527 } 4528 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4529 thread->mLock.unlock(); 4530 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4531 thread->mLock.lock(); 4532 4533#ifdef ADD_BATTERY_DATA 4534 // to track the speaker usage 4535 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4536#endif 4537 } 4538 } 4539} 4540 4541void AudioFlinger::PlaybackThread::Track::pause() 4542{ 4543 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4544 sp<ThreadBase> thread = mThread.promote(); 4545 if (thread != 0) { 4546 Mutex::Autolock _l(thread->mLock); 4547 if (mState == ACTIVE || mState == RESUMING) { 4548 mState = PAUSING; 4549 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4550 if (!isOutputTrack()) { 4551 thread->mLock.unlock(); 4552 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4553 thread->mLock.lock(); 4554 4555#ifdef ADD_BATTERY_DATA 4556 // to track the speaker usage 4557 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4558#endif 4559 } 4560 } 4561 } 4562} 4563 4564void AudioFlinger::PlaybackThread::Track::flush() 4565{ 4566 ALOGV("flush(%d)", mName); 4567 sp<ThreadBase> thread = mThread.promote(); 4568 if (thread != 0) { 4569 Mutex::Autolock _l(thread->mLock); 4570 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4571 mState != PAUSING) { 4572 return; 4573 } 4574 // No point remaining in PAUSED state after a flush => go to 4575 // FLUSHED state 4576 mState = FLUSHED; 4577 // do not reset the track if it is still in the process of being stopped or paused. 4578 // this will be done by prepareTracks_l() when the track is stopped. 4579 // prepareTracks_l() will see mState == FLUSHED, then 4580 // remove from active track list, reset(), and trigger presentation complete 4581 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4582 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4583 reset(); 4584 } 4585 } 4586} 4587 4588void AudioFlinger::PlaybackThread::Track::reset() 4589{ 4590 // Do not reset twice to avoid discarding data written just after a flush and before 4591 // the audioflinger thread detects the track is stopped. 4592 if (!mResetDone) { 4593 TrackBase::reset(); 4594 // Force underrun condition to avoid false underrun callback until first data is 4595 // written to buffer 4596 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4597 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4598 mFillingUpStatus = FS_FILLING; 4599 mResetDone = true; 4600 if (mState == FLUSHED) { 4601 mState = IDLE; 4602 } 4603 } 4604} 4605 4606void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4607{ 4608 mMute = muted; 4609} 4610 4611status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4612{ 4613 status_t status = DEAD_OBJECT; 4614 sp<ThreadBase> thread = mThread.promote(); 4615 if (thread != 0) { 4616 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4617 status = playbackThread->attachAuxEffect(this, EffectId); 4618 } 4619 return status; 4620} 4621 4622void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4623{ 4624 mAuxEffectId = EffectId; 4625 mAuxBuffer = buffer; 4626} 4627 4628bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4629 size_t audioHalFrames) 4630{ 4631 // a track is considered presented when the total number of frames written to audio HAL 4632 // corresponds to the number of frames written when presentationComplete() is called for the 4633 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4634 if (mPresentationCompleteFrames == 0) { 4635 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4636 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4637 mPresentationCompleteFrames, audioHalFrames); 4638 } 4639 if (framesWritten >= mPresentationCompleteFrames) { 4640 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4641 mSessionId, framesWritten); 4642 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4643 return true; 4644 } 4645 return false; 4646} 4647 4648void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4649{ 4650 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4651 if (mSyncEvents[i]->type() == type) { 4652 mSyncEvents[i]->trigger(); 4653 mSyncEvents.removeAt(i); 4654 i--; 4655 } 4656 } 4657} 4658 4659// implement VolumeBufferProvider interface 4660 4661uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4662{ 4663 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4664 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4665 uint32_t vlr = mCblk->getVolumeLR(); 4666 uint32_t vl = vlr & 0xFFFF; 4667 uint32_t vr = vlr >> 16; 4668 // track volumes come from shared memory, so can't be trusted and must be clamped 4669 if (vl > MAX_GAIN_INT) { 4670 vl = MAX_GAIN_INT; 4671 } 4672 if (vr > MAX_GAIN_INT) { 4673 vr = MAX_GAIN_INT; 4674 } 4675 // now apply the cached master volume and stream type volume; 4676 // this is trusted but lacks any synchronization or barrier so may be stale 4677 float v = mCachedVolume; 4678 vl *= v; 4679 vr *= v; 4680 // re-combine into U4.16 4681 vlr = (vr << 16) | (vl & 0xFFFF); 4682 // FIXME look at mute, pause, and stop flags 4683 return vlr; 4684} 4685 4686status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4687{ 4688 if (mState == TERMINATED || mState == PAUSED || 4689 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4690 (mState == STOPPED)))) { 4691 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4692 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4693 event->cancel(); 4694 return INVALID_OPERATION; 4695 } 4696 TrackBase::setSyncEvent(event); 4697 return NO_ERROR; 4698} 4699 4700// timed audio tracks 4701 4702sp<AudioFlinger::PlaybackThread::TimedTrack> 4703AudioFlinger::PlaybackThread::TimedTrack::create( 4704 PlaybackThread *thread, 4705 const sp<Client>& client, 4706 audio_stream_type_t streamType, 4707 uint32_t sampleRate, 4708 audio_format_t format, 4709 uint32_t channelMask, 4710 int frameCount, 4711 const sp<IMemory>& sharedBuffer, 4712 int sessionId) { 4713 if (!client->reserveTimedTrack()) 4714 return NULL; 4715 4716 return new TimedTrack( 4717 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4718 sharedBuffer, sessionId); 4719} 4720 4721AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4722 PlaybackThread *thread, 4723 const sp<Client>& client, 4724 audio_stream_type_t streamType, 4725 uint32_t sampleRate, 4726 audio_format_t format, 4727 uint32_t channelMask, 4728 int frameCount, 4729 const sp<IMemory>& sharedBuffer, 4730 int sessionId) 4731 : Track(thread, client, streamType, sampleRate, format, channelMask, 4732 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4733 mQueueHeadInFlight(false), 4734 mTrimQueueHeadOnRelease(false), 4735 mFramesPendingInQueue(0), 4736 mTimedSilenceBuffer(NULL), 4737 mTimedSilenceBufferSize(0), 4738 mTimedAudioOutputOnTime(false), 4739 mMediaTimeTransformValid(false) 4740{ 4741 LocalClock lc; 4742 mLocalTimeFreq = lc.getLocalFreq(); 4743 4744 mLocalTimeToSampleTransform.a_zero = 0; 4745 mLocalTimeToSampleTransform.b_zero = 0; 4746 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4747 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4748 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4749 &mLocalTimeToSampleTransform.a_to_b_denom); 4750 4751 mMediaTimeToSampleTransform.a_zero = 0; 4752 mMediaTimeToSampleTransform.b_zero = 0; 4753 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4754 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4755 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4756 &mMediaTimeToSampleTransform.a_to_b_denom); 4757} 4758 4759AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4760 mClient->releaseTimedTrack(); 4761 delete [] mTimedSilenceBuffer; 4762} 4763 4764status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4765 size_t size, sp<IMemory>* buffer) { 4766 4767 Mutex::Autolock _l(mTimedBufferQueueLock); 4768 4769 trimTimedBufferQueue_l(); 4770 4771 // lazily initialize the shared memory heap for timed buffers 4772 if (mTimedMemoryDealer == NULL) { 4773 const int kTimedBufferHeapSize = 512 << 10; 4774 4775 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4776 "AudioFlingerTimed"); 4777 if (mTimedMemoryDealer == NULL) 4778 return NO_MEMORY; 4779 } 4780 4781 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4782 if (newBuffer == NULL) { 4783 newBuffer = mTimedMemoryDealer->allocate(size); 4784 if (newBuffer == NULL) 4785 return NO_MEMORY; 4786 } 4787 4788 *buffer = newBuffer; 4789 return NO_ERROR; 4790} 4791 4792// caller must hold mTimedBufferQueueLock 4793void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4794 int64_t mediaTimeNow; 4795 { 4796 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4797 if (!mMediaTimeTransformValid) 4798 return; 4799 4800 int64_t targetTimeNow; 4801 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4802 ? mCCHelper.getCommonTime(&targetTimeNow) 4803 : mCCHelper.getLocalTime(&targetTimeNow); 4804 4805 if (OK != res) 4806 return; 4807 4808 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4809 &mediaTimeNow)) { 4810 return; 4811 } 4812 } 4813 4814 size_t trimEnd; 4815 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4816 int64_t bufEnd; 4817 4818 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4819 // We have a next buffer. Just use its PTS as the PTS of the frame 4820 // following the last frame in this buffer. If the stream is sparse 4821 // (ie, there are deliberate gaps left in the stream which should be 4822 // filled with silence by the TimedAudioTrack), then this can result 4823 // in one extra buffer being left un-trimmed when it could have 4824 // been. In general, this is not typical, and we would rather 4825 // optimized away the TS calculation below for the more common case 4826 // where PTSes are contiguous. 4827 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4828 } else { 4829 // We have no next buffer. Compute the PTS of the frame following 4830 // the last frame in this buffer by computing the duration of of 4831 // this frame in media time units and adding it to the PTS of the 4832 // buffer. 4833 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4834 / mCblk->frameSize; 4835 4836 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4837 &bufEnd)) { 4838 ALOGE("Failed to convert frame count of %lld to media time" 4839 " duration" " (scale factor %d/%u) in %s", 4840 frameCount, 4841 mMediaTimeToSampleTransform.a_to_b_numer, 4842 mMediaTimeToSampleTransform.a_to_b_denom, 4843 __PRETTY_FUNCTION__); 4844 break; 4845 } 4846 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4847 } 4848 4849 if (bufEnd > mediaTimeNow) 4850 break; 4851 4852 // Is the buffer we want to use in the middle of a mix operation right 4853 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4854 // from the mixer which should be coming back shortly. 4855 if (!trimEnd && mQueueHeadInFlight) { 4856 mTrimQueueHeadOnRelease = true; 4857 } 4858 } 4859 4860 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4861 if (trimStart < trimEnd) { 4862 // Update the bookkeeping for framesReady() 4863 for (size_t i = trimStart; i < trimEnd; ++i) { 4864 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4865 } 4866 4867 // Now actually remove the buffers from the queue. 4868 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4869 } 4870} 4871 4872void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4873 const char* logTag) { 4874 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4875 "%s called (reason \"%s\"), but timed buffer queue has no" 4876 " elements to trim.", __FUNCTION__, logTag); 4877 4878 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4879 mTimedBufferQueue.removeAt(0); 4880} 4881 4882void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4883 const TimedBuffer& buf, 4884 const char* logTag) { 4885 uint32_t bufBytes = buf.buffer()->size(); 4886 uint32_t consumedAlready = buf.position(); 4887 4888 ALOG_ASSERT(consumedAlready <= bufBytes, 4889 "Bad bookkeeping while updating frames pending. Timed buffer is" 4890 " only %u bytes long, but claims to have consumed %u" 4891 " bytes. (update reason: \"%s\")", 4892 bufBytes, consumedAlready, logTag); 4893 4894 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4895 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4896 "Bad bookkeeping while updating frames pending. Should have at" 4897 " least %u queued frames, but we think we have only %u. (update" 4898 " reason: \"%s\")", 4899 bufFrames, mFramesPendingInQueue, logTag); 4900 4901 mFramesPendingInQueue -= bufFrames; 4902} 4903 4904status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4905 const sp<IMemory>& buffer, int64_t pts) { 4906 4907 { 4908 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4909 if (!mMediaTimeTransformValid) 4910 return INVALID_OPERATION; 4911 } 4912 4913 Mutex::Autolock _l(mTimedBufferQueueLock); 4914 4915 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4916 mFramesPendingInQueue += bufFrames; 4917 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4918 4919 return NO_ERROR; 4920} 4921 4922status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4923 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4924 4925 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4926 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4927 target); 4928 4929 if (!(target == TimedAudioTrack::LOCAL_TIME || 4930 target == TimedAudioTrack::COMMON_TIME)) { 4931 return BAD_VALUE; 4932 } 4933 4934 Mutex::Autolock lock(mMediaTimeTransformLock); 4935 mMediaTimeTransform = xform; 4936 mMediaTimeTransformTarget = target; 4937 mMediaTimeTransformValid = true; 4938 4939 return NO_ERROR; 4940} 4941 4942#define min(a, b) ((a) < (b) ? (a) : (b)) 4943 4944// implementation of getNextBuffer for tracks whose buffers have timestamps 4945status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4946 AudioBufferProvider::Buffer* buffer, int64_t pts) 4947{ 4948 if (pts == AudioBufferProvider::kInvalidPTS) { 4949 buffer->raw = 0; 4950 buffer->frameCount = 0; 4951 mTimedAudioOutputOnTime = false; 4952 return INVALID_OPERATION; 4953 } 4954 4955 Mutex::Autolock _l(mTimedBufferQueueLock); 4956 4957 ALOG_ASSERT(!mQueueHeadInFlight, 4958 "getNextBuffer called without releaseBuffer!"); 4959 4960 while (true) { 4961 4962 // if we have no timed buffers, then fail 4963 if (mTimedBufferQueue.isEmpty()) { 4964 buffer->raw = 0; 4965 buffer->frameCount = 0; 4966 return NOT_ENOUGH_DATA; 4967 } 4968 4969 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4970 4971 // calculate the PTS of the head of the timed buffer queue expressed in 4972 // local time 4973 int64_t headLocalPTS; 4974 { 4975 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4976 4977 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4978 4979 if (mMediaTimeTransform.a_to_b_denom == 0) { 4980 // the transform represents a pause, so yield silence 4981 timedYieldSilence_l(buffer->frameCount, buffer); 4982 return NO_ERROR; 4983 } 4984 4985 int64_t transformedPTS; 4986 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4987 &transformedPTS)) { 4988 // the transform failed. this shouldn't happen, but if it does 4989 // then just drop this buffer 4990 ALOGW("timedGetNextBuffer transform failed"); 4991 buffer->raw = 0; 4992 buffer->frameCount = 0; 4993 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 4994 return NO_ERROR; 4995 } 4996 4997 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4998 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4999 &headLocalPTS)) { 5000 buffer->raw = 0; 5001 buffer->frameCount = 0; 5002 return INVALID_OPERATION; 5003 } 5004 } else { 5005 headLocalPTS = transformedPTS; 5006 } 5007 } 5008 5009 // adjust the head buffer's PTS to reflect the portion of the head buffer 5010 // that has already been consumed 5011 int64_t effectivePTS = headLocalPTS + 5012 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5013 5014 // Calculate the delta in samples between the head of the input buffer 5015 // queue and the start of the next output buffer that will be written. 5016 // If the transformation fails because of over or underflow, it means 5017 // that the sample's position in the output stream is so far out of 5018 // whack that it should just be dropped. 5019 int64_t sampleDelta; 5020 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5021 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5022 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5023 " mix"); 5024 continue; 5025 } 5026 if (!mLocalTimeToSampleTransform.doForwardTransform( 5027 (effectivePTS - pts) << 32, &sampleDelta)) { 5028 ALOGV("*** too late during sample rate transform: dropped buffer"); 5029 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5030 continue; 5031 } 5032 5033 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5034 " sampleDelta=[%d.%08x]", 5035 head.pts(), head.position(), pts, 5036 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5037 + (sampleDelta >> 32)), 5038 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5039 5040 // if the delta between the ideal placement for the next input sample and 5041 // the current output position is within this threshold, then we will 5042 // concatenate the next input samples to the previous output 5043 const int64_t kSampleContinuityThreshold = 5044 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5045 5046 // if this is the first buffer of audio that we're emitting from this track 5047 // then it should be almost exactly on time. 5048 const int64_t kSampleStartupThreshold = 1LL << 32; 5049 5050 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5051 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5052 // the next input is close enough to being on time, so concatenate it 5053 // with the last output 5054 timedYieldSamples_l(buffer); 5055 5056 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5057 head.position(), buffer->frameCount); 5058 return NO_ERROR; 5059 } 5060 5061 // Looks like our output is not on time. Reset our on timed status. 5062 // Next time we mix samples from our input queue, then should be within 5063 // the StartupThreshold. 5064 mTimedAudioOutputOnTime = false; 5065 if (sampleDelta > 0) { 5066 // the gap between the current output position and the proper start of 5067 // the next input sample is too big, so fill it with silence 5068 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5069 5070 timedYieldSilence_l(framesUntilNextInput, buffer); 5071 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5072 return NO_ERROR; 5073 } else { 5074 // the next input sample is late 5075 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5076 size_t onTimeSamplePosition = 5077 head.position() + lateFrames * mCblk->frameSize; 5078 5079 if (onTimeSamplePosition > head.buffer()->size()) { 5080 // all the remaining samples in the head are too late, so 5081 // drop it and move on 5082 ALOGV("*** too late: dropped buffer"); 5083 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5084 continue; 5085 } else { 5086 // skip over the late samples 5087 head.setPosition(onTimeSamplePosition); 5088 5089 // yield the available samples 5090 timedYieldSamples_l(buffer); 5091 5092 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5093 return NO_ERROR; 5094 } 5095 } 5096 } 5097} 5098 5099// Yield samples from the timed buffer queue head up to the given output 5100// buffer's capacity. 5101// 5102// Caller must hold mTimedBufferQueueLock 5103void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5104 AudioBufferProvider::Buffer* buffer) { 5105 5106 const TimedBuffer& head = mTimedBufferQueue[0]; 5107 5108 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5109 head.position()); 5110 5111 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5112 mCblk->frameSize); 5113 size_t framesRequested = buffer->frameCount; 5114 buffer->frameCount = min(framesLeftInHead, framesRequested); 5115 5116 mQueueHeadInFlight = true; 5117 mTimedAudioOutputOnTime = true; 5118} 5119 5120// Yield samples of silence up to the given output buffer's capacity 5121// 5122// Caller must hold mTimedBufferQueueLock 5123void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5124 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5125 5126 // lazily allocate a buffer filled with silence 5127 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5128 delete [] mTimedSilenceBuffer; 5129 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5130 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5131 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5132 } 5133 5134 buffer->raw = mTimedSilenceBuffer; 5135 size_t framesRequested = buffer->frameCount; 5136 buffer->frameCount = min(numFrames, framesRequested); 5137 5138 mTimedAudioOutputOnTime = false; 5139} 5140 5141// AudioBufferProvider interface 5142void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5143 AudioBufferProvider::Buffer* buffer) { 5144 5145 Mutex::Autolock _l(mTimedBufferQueueLock); 5146 5147 // If the buffer which was just released is part of the buffer at the head 5148 // of the queue, be sure to update the amt of the buffer which has been 5149 // consumed. If the buffer being returned is not part of the head of the 5150 // queue, its either because the buffer is part of the silence buffer, or 5151 // because the head of the timed queue was trimmed after the mixer called 5152 // getNextBuffer but before the mixer called releaseBuffer. 5153 if (buffer->raw == mTimedSilenceBuffer) { 5154 ALOG_ASSERT(!mQueueHeadInFlight, 5155 "Queue head in flight during release of silence buffer!"); 5156 goto done; 5157 } 5158 5159 ALOG_ASSERT(mQueueHeadInFlight, 5160 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5161 " head in flight."); 5162 5163 if (mTimedBufferQueue.size()) { 5164 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5165 5166 void* start = head.buffer()->pointer(); 5167 void* end = reinterpret_cast<void*>( 5168 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5169 + head.buffer()->size()); 5170 5171 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5172 "released buffer not within the head of the timed buffer" 5173 " queue; qHead = [%p, %p], released buffer = %p", 5174 start, end, buffer->raw); 5175 5176 head.setPosition(head.position() + 5177 (buffer->frameCount * mCblk->frameSize)); 5178 mQueueHeadInFlight = false; 5179 5180 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5181 "Bad bookkeeping during releaseBuffer! Should have at" 5182 " least %u queued frames, but we think we have only %u", 5183 buffer->frameCount, mFramesPendingInQueue); 5184 5185 mFramesPendingInQueue -= buffer->frameCount; 5186 5187 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5188 || mTrimQueueHeadOnRelease) { 5189 trimTimedBufferQueueHead_l("releaseBuffer"); 5190 mTrimQueueHeadOnRelease = false; 5191 } 5192 } else { 5193 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5194 " buffers in the timed buffer queue"); 5195 } 5196 5197done: 5198 buffer->raw = 0; 5199 buffer->frameCount = 0; 5200} 5201 5202size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5203 Mutex::Autolock _l(mTimedBufferQueueLock); 5204 return mFramesPendingInQueue; 5205} 5206 5207AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5208 : mPTS(0), mPosition(0) {} 5209 5210AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5211 const sp<IMemory>& buffer, int64_t pts) 5212 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5213 5214// ---------------------------------------------------------------------------- 5215 5216// RecordTrack constructor must be called with AudioFlinger::mLock held 5217AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5218 RecordThread *thread, 5219 const sp<Client>& client, 5220 uint32_t sampleRate, 5221 audio_format_t format, 5222 uint32_t channelMask, 5223 int frameCount, 5224 int sessionId) 5225 : TrackBase(thread, client, sampleRate, format, 5226 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5227 mOverflow(false) 5228{ 5229 if (mCblk != NULL) { 5230 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5231 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5232 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5233 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5234 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5235 } else { 5236 mCblk->frameSize = sizeof(int8_t); 5237 } 5238 } 5239} 5240 5241AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5242{ 5243 sp<ThreadBase> thread = mThread.promote(); 5244 if (thread != 0) { 5245 AudioSystem::releaseInput(thread->id()); 5246 } 5247} 5248 5249// AudioBufferProvider interface 5250status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5251{ 5252 audio_track_cblk_t* cblk = this->cblk(); 5253 uint32_t framesAvail; 5254 uint32_t framesReq = buffer->frameCount; 5255 5256 // Check if last stepServer failed, try to step now 5257 if (mStepServerFailed) { 5258 if (!step()) goto getNextBuffer_exit; 5259 ALOGV("stepServer recovered"); 5260 mStepServerFailed = false; 5261 } 5262 5263 framesAvail = cblk->framesAvailable_l(); 5264 5265 if (CC_LIKELY(framesAvail)) { 5266 uint32_t s = cblk->server; 5267 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5268 5269 if (framesReq > framesAvail) { 5270 framesReq = framesAvail; 5271 } 5272 if (framesReq > bufferEnd - s) { 5273 framesReq = bufferEnd - s; 5274 } 5275 5276 buffer->raw = getBuffer(s, framesReq); 5277 if (buffer->raw == NULL) goto getNextBuffer_exit; 5278 5279 buffer->frameCount = framesReq; 5280 return NO_ERROR; 5281 } 5282 5283getNextBuffer_exit: 5284 buffer->raw = NULL; 5285 buffer->frameCount = 0; 5286 return NOT_ENOUGH_DATA; 5287} 5288 5289status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5290 int triggerSession) 5291{ 5292 sp<ThreadBase> thread = mThread.promote(); 5293 if (thread != 0) { 5294 RecordThread *recordThread = (RecordThread *)thread.get(); 5295 return recordThread->start(this, event, triggerSession); 5296 } else { 5297 return BAD_VALUE; 5298 } 5299} 5300 5301void AudioFlinger::RecordThread::RecordTrack::stop() 5302{ 5303 sp<ThreadBase> thread = mThread.promote(); 5304 if (thread != 0) { 5305 RecordThread *recordThread = (RecordThread *)thread.get(); 5306 recordThread->stop(this); 5307 TrackBase::reset(); 5308 // Force overrun condition to avoid false overrun callback until first data is 5309 // read from buffer 5310 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5311 } 5312} 5313 5314void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5315{ 5316 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5317 (mClient == 0) ? getpid_cached : mClient->pid(), 5318 mFormat, 5319 mChannelMask, 5320 mSessionId, 5321 mFrameCount, 5322 mState, 5323 mCblk->sampleRate, 5324 mCblk->server, 5325 mCblk->user); 5326} 5327 5328 5329// ---------------------------------------------------------------------------- 5330 5331AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5332 PlaybackThread *playbackThread, 5333 DuplicatingThread *sourceThread, 5334 uint32_t sampleRate, 5335 audio_format_t format, 5336 uint32_t channelMask, 5337 int frameCount) 5338 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5339 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5340 mActive(false), mSourceThread(sourceThread) 5341{ 5342 5343 if (mCblk != NULL) { 5344 mCblk->flags |= CBLK_DIRECTION_OUT; 5345 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5346 mOutBuffer.frameCount = 0; 5347 playbackThread->mTracks.add(this); 5348 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5349 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5350 mCblk, mBuffer, mCblk->buffers, 5351 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5352 } else { 5353 ALOGW("Error creating output track on thread %p", playbackThread); 5354 } 5355} 5356 5357AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5358{ 5359 clearBufferQueue(); 5360} 5361 5362status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5363 int triggerSession) 5364{ 5365 status_t status = Track::start(event, triggerSession); 5366 if (status != NO_ERROR) { 5367 return status; 5368 } 5369 5370 mActive = true; 5371 mRetryCount = 127; 5372 return status; 5373} 5374 5375void AudioFlinger::PlaybackThread::OutputTrack::stop() 5376{ 5377 Track::stop(); 5378 clearBufferQueue(); 5379 mOutBuffer.frameCount = 0; 5380 mActive = false; 5381} 5382 5383bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5384{ 5385 Buffer *pInBuffer; 5386 Buffer inBuffer; 5387 uint32_t channelCount = mChannelCount; 5388 bool outputBufferFull = false; 5389 inBuffer.frameCount = frames; 5390 inBuffer.i16 = data; 5391 5392 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5393 5394 if (!mActive && frames != 0) { 5395 start(); 5396 sp<ThreadBase> thread = mThread.promote(); 5397 if (thread != 0) { 5398 MixerThread *mixerThread = (MixerThread *)thread.get(); 5399 if (mCblk->frameCount > frames){ 5400 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5401 uint32_t startFrames = (mCblk->frameCount - frames); 5402 pInBuffer = new Buffer; 5403 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5404 pInBuffer->frameCount = startFrames; 5405 pInBuffer->i16 = pInBuffer->mBuffer; 5406 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5407 mBufferQueue.add(pInBuffer); 5408 } else { 5409 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5410 } 5411 } 5412 } 5413 } 5414 5415 while (waitTimeLeftMs) { 5416 // First write pending buffers, then new data 5417 if (mBufferQueue.size()) { 5418 pInBuffer = mBufferQueue.itemAt(0); 5419 } else { 5420 pInBuffer = &inBuffer; 5421 } 5422 5423 if (pInBuffer->frameCount == 0) { 5424 break; 5425 } 5426 5427 if (mOutBuffer.frameCount == 0) { 5428 mOutBuffer.frameCount = pInBuffer->frameCount; 5429 nsecs_t startTime = systemTime(); 5430 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5431 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5432 outputBufferFull = true; 5433 break; 5434 } 5435 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5436 if (waitTimeLeftMs >= waitTimeMs) { 5437 waitTimeLeftMs -= waitTimeMs; 5438 } else { 5439 waitTimeLeftMs = 0; 5440 } 5441 } 5442 5443 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5444 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5445 mCblk->stepUser(outFrames); 5446 pInBuffer->frameCount -= outFrames; 5447 pInBuffer->i16 += outFrames * channelCount; 5448 mOutBuffer.frameCount -= outFrames; 5449 mOutBuffer.i16 += outFrames * channelCount; 5450 5451 if (pInBuffer->frameCount == 0) { 5452 if (mBufferQueue.size()) { 5453 mBufferQueue.removeAt(0); 5454 delete [] pInBuffer->mBuffer; 5455 delete pInBuffer; 5456 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5457 } else { 5458 break; 5459 } 5460 } 5461 } 5462 5463 // If we could not write all frames, allocate a buffer and queue it for next time. 5464 if (inBuffer.frameCount) { 5465 sp<ThreadBase> thread = mThread.promote(); 5466 if (thread != 0 && !thread->standby()) { 5467 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5468 pInBuffer = new Buffer; 5469 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5470 pInBuffer->frameCount = inBuffer.frameCount; 5471 pInBuffer->i16 = pInBuffer->mBuffer; 5472 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5473 mBufferQueue.add(pInBuffer); 5474 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5475 } else { 5476 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5477 } 5478 } 5479 } 5480 5481 // Calling write() with a 0 length buffer, means that no more data will be written: 5482 // If no more buffers are pending, fill output track buffer to make sure it is started 5483 // by output mixer. 5484 if (frames == 0 && mBufferQueue.size() == 0) { 5485 if (mCblk->user < mCblk->frameCount) { 5486 frames = mCblk->frameCount - mCblk->user; 5487 pInBuffer = new Buffer; 5488 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5489 pInBuffer->frameCount = frames; 5490 pInBuffer->i16 = pInBuffer->mBuffer; 5491 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5492 mBufferQueue.add(pInBuffer); 5493 } else if (mActive) { 5494 stop(); 5495 } 5496 } 5497 5498 return outputBufferFull; 5499} 5500 5501status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5502{ 5503 int active; 5504 status_t result; 5505 audio_track_cblk_t* cblk = mCblk; 5506 uint32_t framesReq = buffer->frameCount; 5507 5508// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5509 buffer->frameCount = 0; 5510 5511 uint32_t framesAvail = cblk->framesAvailable(); 5512 5513 5514 if (framesAvail == 0) { 5515 Mutex::Autolock _l(cblk->lock); 5516 goto start_loop_here; 5517 while (framesAvail == 0) { 5518 active = mActive; 5519 if (CC_UNLIKELY(!active)) { 5520 ALOGV("Not active and NO_MORE_BUFFERS"); 5521 return NO_MORE_BUFFERS; 5522 } 5523 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5524 if (result != NO_ERROR) { 5525 return NO_MORE_BUFFERS; 5526 } 5527 // read the server count again 5528 start_loop_here: 5529 framesAvail = cblk->framesAvailable_l(); 5530 } 5531 } 5532 5533// if (framesAvail < framesReq) { 5534// return NO_MORE_BUFFERS; 5535// } 5536 5537 if (framesReq > framesAvail) { 5538 framesReq = framesAvail; 5539 } 5540 5541 uint32_t u = cblk->user; 5542 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5543 5544 if (framesReq > bufferEnd - u) { 5545 framesReq = bufferEnd - u; 5546 } 5547 5548 buffer->frameCount = framesReq; 5549 buffer->raw = (void *)cblk->buffer(u); 5550 return NO_ERROR; 5551} 5552 5553 5554void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5555{ 5556 size_t size = mBufferQueue.size(); 5557 5558 for (size_t i = 0; i < size; i++) { 5559 Buffer *pBuffer = mBufferQueue.itemAt(i); 5560 delete [] pBuffer->mBuffer; 5561 delete pBuffer; 5562 } 5563 mBufferQueue.clear(); 5564} 5565 5566// ---------------------------------------------------------------------------- 5567 5568AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5569 : RefBase(), 5570 mAudioFlinger(audioFlinger), 5571 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5572 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5573 mPid(pid), 5574 mTimedTrackCount(0) 5575{ 5576 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5577} 5578 5579// Client destructor must be called with AudioFlinger::mLock held 5580AudioFlinger::Client::~Client() 5581{ 5582 mAudioFlinger->removeClient_l(mPid); 5583} 5584 5585sp<MemoryDealer> AudioFlinger::Client::heap() const 5586{ 5587 return mMemoryDealer; 5588} 5589 5590// Reserve one of the limited slots for a timed audio track associated 5591// with this client 5592bool AudioFlinger::Client::reserveTimedTrack() 5593{ 5594 const int kMaxTimedTracksPerClient = 4; 5595 5596 Mutex::Autolock _l(mTimedTrackLock); 5597 5598 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5599 ALOGW("can not create timed track - pid %d has exceeded the limit", 5600 mPid); 5601 return false; 5602 } 5603 5604 mTimedTrackCount++; 5605 return true; 5606} 5607 5608// Release a slot for a timed audio track 5609void AudioFlinger::Client::releaseTimedTrack() 5610{ 5611 Mutex::Autolock _l(mTimedTrackLock); 5612 mTimedTrackCount--; 5613} 5614 5615// ---------------------------------------------------------------------------- 5616 5617AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5618 const sp<IAudioFlingerClient>& client, 5619 pid_t pid) 5620 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5621{ 5622} 5623 5624AudioFlinger::NotificationClient::~NotificationClient() 5625{ 5626} 5627 5628void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5629{ 5630 sp<NotificationClient> keep(this); 5631 mAudioFlinger->removeNotificationClient(mPid); 5632} 5633 5634// ---------------------------------------------------------------------------- 5635 5636AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5637 : BnAudioTrack(), 5638 mTrack(track) 5639{ 5640} 5641 5642AudioFlinger::TrackHandle::~TrackHandle() { 5643 // just stop the track on deletion, associated resources 5644 // will be freed from the main thread once all pending buffers have 5645 // been played. Unless it's not in the active track list, in which 5646 // case we free everything now... 5647 mTrack->destroy(); 5648} 5649 5650sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5651 return mTrack->getCblk(); 5652} 5653 5654status_t AudioFlinger::TrackHandle::start() { 5655 return mTrack->start(); 5656} 5657 5658void AudioFlinger::TrackHandle::stop() { 5659 mTrack->stop(); 5660} 5661 5662void AudioFlinger::TrackHandle::flush() { 5663 mTrack->flush(); 5664} 5665 5666void AudioFlinger::TrackHandle::mute(bool e) { 5667 mTrack->mute(e); 5668} 5669 5670void AudioFlinger::TrackHandle::pause() { 5671 mTrack->pause(); 5672} 5673 5674status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5675{ 5676 return mTrack->attachAuxEffect(EffectId); 5677} 5678 5679status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5680 sp<IMemory>* buffer) { 5681 if (!mTrack->isTimedTrack()) 5682 return INVALID_OPERATION; 5683 5684 PlaybackThread::TimedTrack* tt = 5685 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5686 return tt->allocateTimedBuffer(size, buffer); 5687} 5688 5689status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5690 int64_t pts) { 5691 if (!mTrack->isTimedTrack()) 5692 return INVALID_OPERATION; 5693 5694 PlaybackThread::TimedTrack* tt = 5695 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5696 return tt->queueTimedBuffer(buffer, pts); 5697} 5698 5699status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5700 const LinearTransform& xform, int target) { 5701 5702 if (!mTrack->isTimedTrack()) 5703 return INVALID_OPERATION; 5704 5705 PlaybackThread::TimedTrack* tt = 5706 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5707 return tt->setMediaTimeTransform( 5708 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5709} 5710 5711status_t AudioFlinger::TrackHandle::onTransact( 5712 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5713{ 5714 return BnAudioTrack::onTransact(code, data, reply, flags); 5715} 5716 5717// ---------------------------------------------------------------------------- 5718 5719sp<IAudioRecord> AudioFlinger::openRecord( 5720 pid_t pid, 5721 audio_io_handle_t input, 5722 uint32_t sampleRate, 5723 audio_format_t format, 5724 uint32_t channelMask, 5725 int frameCount, 5726 IAudioFlinger::track_flags_t flags, 5727 int *sessionId, 5728 status_t *status) 5729{ 5730 sp<RecordThread::RecordTrack> recordTrack; 5731 sp<RecordHandle> recordHandle; 5732 sp<Client> client; 5733 status_t lStatus; 5734 RecordThread *thread; 5735 size_t inFrameCount; 5736 int lSessionId; 5737 5738 // check calling permissions 5739 if (!recordingAllowed()) { 5740 lStatus = PERMISSION_DENIED; 5741 goto Exit; 5742 } 5743 5744 // add client to list 5745 { // scope for mLock 5746 Mutex::Autolock _l(mLock); 5747 thread = checkRecordThread_l(input); 5748 if (thread == NULL) { 5749 lStatus = BAD_VALUE; 5750 goto Exit; 5751 } 5752 5753 client = registerPid_l(pid); 5754 5755 // If no audio session id is provided, create one here 5756 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5757 lSessionId = *sessionId; 5758 } else { 5759 lSessionId = nextUniqueId(); 5760 if (sessionId != NULL) { 5761 *sessionId = lSessionId; 5762 } 5763 } 5764 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5765 recordTrack = thread->createRecordTrack_l(client, 5766 sampleRate, 5767 format, 5768 channelMask, 5769 frameCount, 5770 lSessionId, 5771 &lStatus); 5772 } 5773 if (lStatus != NO_ERROR) { 5774 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5775 // destructor is called by the TrackBase destructor with mLock held 5776 client.clear(); 5777 recordTrack.clear(); 5778 goto Exit; 5779 } 5780 5781 // return to handle to client 5782 recordHandle = new RecordHandle(recordTrack); 5783 lStatus = NO_ERROR; 5784 5785Exit: 5786 if (status) { 5787 *status = lStatus; 5788 } 5789 return recordHandle; 5790} 5791 5792// ---------------------------------------------------------------------------- 5793 5794AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5795 : BnAudioRecord(), 5796 mRecordTrack(recordTrack) 5797{ 5798} 5799 5800AudioFlinger::RecordHandle::~RecordHandle() { 5801 stop(); 5802} 5803 5804sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5805 return mRecordTrack->getCblk(); 5806} 5807 5808status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5809 ALOGV("RecordHandle::start()"); 5810 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5811} 5812 5813void AudioFlinger::RecordHandle::stop() { 5814 ALOGV("RecordHandle::stop()"); 5815 mRecordTrack->stop(); 5816} 5817 5818status_t AudioFlinger::RecordHandle::onTransact( 5819 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5820{ 5821 return BnAudioRecord::onTransact(code, data, reply, flags); 5822} 5823 5824// ---------------------------------------------------------------------------- 5825 5826AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5827 AudioStreamIn *input, 5828 uint32_t sampleRate, 5829 uint32_t channels, 5830 audio_io_handle_t id, 5831 uint32_t device) : 5832 ThreadBase(audioFlinger, id, device, RECORD), 5833 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5834 // mRsmpInIndex and mInputBytes set by readInputParameters() 5835 mReqChannelCount(popcount(channels)), 5836 mReqSampleRate(sampleRate) 5837 // mBytesRead is only meaningful while active, and so is cleared in start() 5838 // (but might be better to also clear here for dump?) 5839{ 5840 snprintf(mName, kNameLength, "AudioIn_%X", id); 5841 5842 readInputParameters(); 5843} 5844 5845 5846AudioFlinger::RecordThread::~RecordThread() 5847{ 5848 delete[] mRsmpInBuffer; 5849 delete mResampler; 5850 delete[] mRsmpOutBuffer; 5851} 5852 5853void AudioFlinger::RecordThread::onFirstRef() 5854{ 5855 run(mName, PRIORITY_URGENT_AUDIO); 5856} 5857 5858status_t AudioFlinger::RecordThread::readyToRun() 5859{ 5860 status_t status = initCheck(); 5861 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5862 return status; 5863} 5864 5865bool AudioFlinger::RecordThread::threadLoop() 5866{ 5867 AudioBufferProvider::Buffer buffer; 5868 sp<RecordTrack> activeTrack; 5869 Vector< sp<EffectChain> > effectChains; 5870 5871 nsecs_t lastWarning = 0; 5872 5873 acquireWakeLock(); 5874 5875 // start recording 5876 while (!exitPending()) { 5877 5878 processConfigEvents(); 5879 5880 { // scope for mLock 5881 Mutex::Autolock _l(mLock); 5882 checkForNewParameters_l(); 5883 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5884 if (!mStandby) { 5885 mInput->stream->common.standby(&mInput->stream->common); 5886 mStandby = true; 5887 } 5888 5889 if (exitPending()) break; 5890 5891 releaseWakeLock_l(); 5892 ALOGV("RecordThread: loop stopping"); 5893 // go to sleep 5894 mWaitWorkCV.wait(mLock); 5895 ALOGV("RecordThread: loop starting"); 5896 acquireWakeLock_l(); 5897 continue; 5898 } 5899 if (mActiveTrack != 0) { 5900 if (mActiveTrack->mState == TrackBase::PAUSING) { 5901 if (!mStandby) { 5902 mInput->stream->common.standby(&mInput->stream->common); 5903 mStandby = true; 5904 } 5905 mActiveTrack.clear(); 5906 mStartStopCond.broadcast(); 5907 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5908 if (mReqChannelCount != mActiveTrack->channelCount()) { 5909 mActiveTrack.clear(); 5910 mStartStopCond.broadcast(); 5911 } else if (mBytesRead != 0) { 5912 // record start succeeds only if first read from audio input 5913 // succeeds 5914 if (mBytesRead > 0) { 5915 mActiveTrack->mState = TrackBase::ACTIVE; 5916 } else { 5917 mActiveTrack.clear(); 5918 } 5919 mStartStopCond.broadcast(); 5920 } 5921 mStandby = false; 5922 } 5923 } 5924 lockEffectChains_l(effectChains); 5925 } 5926 5927 if (mActiveTrack != 0) { 5928 if (mActiveTrack->mState != TrackBase::ACTIVE && 5929 mActiveTrack->mState != TrackBase::RESUMING) { 5930 unlockEffectChains(effectChains); 5931 usleep(kRecordThreadSleepUs); 5932 continue; 5933 } 5934 for (size_t i = 0; i < effectChains.size(); i ++) { 5935 effectChains[i]->process_l(); 5936 } 5937 5938 buffer.frameCount = mFrameCount; 5939 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5940 size_t framesOut = buffer.frameCount; 5941 if (mResampler == NULL) { 5942 // no resampling 5943 while (framesOut) { 5944 size_t framesIn = mFrameCount - mRsmpInIndex; 5945 if (framesIn) { 5946 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5947 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5948 if (framesIn > framesOut) 5949 framesIn = framesOut; 5950 mRsmpInIndex += framesIn; 5951 framesOut -= framesIn; 5952 if ((int)mChannelCount == mReqChannelCount || 5953 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5954 memcpy(dst, src, framesIn * mFrameSize); 5955 } else { 5956 int16_t *src16 = (int16_t *)src; 5957 int16_t *dst16 = (int16_t *)dst; 5958 if (mChannelCount == 1) { 5959 while (framesIn--) { 5960 *dst16++ = *src16; 5961 *dst16++ = *src16++; 5962 } 5963 } else { 5964 while (framesIn--) { 5965 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5966 src16 += 2; 5967 } 5968 } 5969 } 5970 } 5971 if (framesOut && mFrameCount == mRsmpInIndex) { 5972 if (framesOut == mFrameCount && 5973 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5974 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5975 framesOut = 0; 5976 } else { 5977 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5978 mRsmpInIndex = 0; 5979 } 5980 if (mBytesRead < 0) { 5981 ALOGE("Error reading audio input"); 5982 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5983 // Force input into standby so that it tries to 5984 // recover at next read attempt 5985 mInput->stream->common.standby(&mInput->stream->common); 5986 usleep(kRecordThreadSleepUs); 5987 } 5988 mRsmpInIndex = mFrameCount; 5989 framesOut = 0; 5990 buffer.frameCount = 0; 5991 } 5992 } 5993 } 5994 } else { 5995 // resampling 5996 5997 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5998 // alter output frame count as if we were expecting stereo samples 5999 if (mChannelCount == 1 && mReqChannelCount == 1) { 6000 framesOut >>= 1; 6001 } 6002 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6003 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6004 // are 32 bit aligned which should be always true. 6005 if (mChannelCount == 2 && mReqChannelCount == 1) { 6006 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6007 // the resampler always outputs stereo samples: do post stereo to mono conversion 6008 int16_t *src = (int16_t *)mRsmpOutBuffer; 6009 int16_t *dst = buffer.i16; 6010 while (framesOut--) { 6011 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6012 src += 2; 6013 } 6014 } else { 6015 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6016 } 6017 6018 } 6019 if (mFramestoDrop == 0) { 6020 mActiveTrack->releaseBuffer(&buffer); 6021 } else { 6022 if (mFramestoDrop > 0) { 6023 mFramestoDrop -= buffer.frameCount; 6024 if (mFramestoDrop <= 0) { 6025 clearSyncStartEvent(); 6026 } 6027 } else { 6028 mFramestoDrop += buffer.frameCount; 6029 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6030 mSyncStartEvent->isCancelled()) { 6031 ALOGW("Synced record %s, session %d, trigger session %d", 6032 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6033 mActiveTrack->sessionId(), 6034 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6035 clearSyncStartEvent(); 6036 } 6037 } 6038 } 6039 mActiveTrack->overflow(); 6040 } 6041 // client isn't retrieving buffers fast enough 6042 else { 6043 if (!mActiveTrack->setOverflow()) { 6044 nsecs_t now = systemTime(); 6045 if ((now - lastWarning) > kWarningThrottleNs) { 6046 ALOGW("RecordThread: buffer overflow"); 6047 lastWarning = now; 6048 } 6049 } 6050 // Release the processor for a while before asking for a new buffer. 6051 // This will give the application more chance to read from the buffer and 6052 // clear the overflow. 6053 usleep(kRecordThreadSleepUs); 6054 } 6055 } 6056 // enable changes in effect chain 6057 unlockEffectChains(effectChains); 6058 effectChains.clear(); 6059 } 6060 6061 if (!mStandby) { 6062 mInput->stream->common.standby(&mInput->stream->common); 6063 } 6064 mActiveTrack.clear(); 6065 6066 mStartStopCond.broadcast(); 6067 6068 releaseWakeLock(); 6069 6070 ALOGV("RecordThread %p exiting", this); 6071 return false; 6072} 6073 6074 6075sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6076 const sp<AudioFlinger::Client>& client, 6077 uint32_t sampleRate, 6078 audio_format_t format, 6079 int channelMask, 6080 int frameCount, 6081 int sessionId, 6082 status_t *status) 6083{ 6084 sp<RecordTrack> track; 6085 status_t lStatus; 6086 6087 lStatus = initCheck(); 6088 if (lStatus != NO_ERROR) { 6089 ALOGE("Audio driver not initialized."); 6090 goto Exit; 6091 } 6092 6093 { // scope for mLock 6094 Mutex::Autolock _l(mLock); 6095 6096 track = new RecordTrack(this, client, sampleRate, 6097 format, channelMask, frameCount, sessionId); 6098 6099 if (track->getCblk() == 0) { 6100 lStatus = NO_MEMORY; 6101 goto Exit; 6102 } 6103 6104 mTrack = track.get(); 6105 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6106 bool suspend = audio_is_bluetooth_sco_device( 6107 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6108 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6109 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6110 } 6111 lStatus = NO_ERROR; 6112 6113Exit: 6114 if (status) { 6115 *status = lStatus; 6116 } 6117 return track; 6118} 6119 6120status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6121 AudioSystem::sync_event_t event, 6122 int triggerSession) 6123{ 6124 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6125 sp<ThreadBase> strongMe = this; 6126 status_t status = NO_ERROR; 6127 6128 if (event == AudioSystem::SYNC_EVENT_NONE) { 6129 clearSyncStartEvent(); 6130 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6131 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6132 triggerSession, 6133 recordTrack->sessionId(), 6134 syncStartEventCallback, 6135 this); 6136 // Sync event can be cancelled by the trigger session if the track is not in a 6137 // compatible state in which case we start record immediately 6138 if (mSyncStartEvent->isCancelled()) { 6139 clearSyncStartEvent(); 6140 } else { 6141 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6142 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6143 } 6144 } 6145 6146 { 6147 AutoMutex lock(mLock); 6148 if (mActiveTrack != 0) { 6149 if (recordTrack != mActiveTrack.get()) { 6150 status = -EBUSY; 6151 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6152 mActiveTrack->mState = TrackBase::ACTIVE; 6153 } 6154 return status; 6155 } 6156 6157 recordTrack->mState = TrackBase::IDLE; 6158 mActiveTrack = recordTrack; 6159 mLock.unlock(); 6160 status_t status = AudioSystem::startInput(mId); 6161 mLock.lock(); 6162 if (status != NO_ERROR) { 6163 mActiveTrack.clear(); 6164 clearSyncStartEvent(); 6165 return status; 6166 } 6167 mRsmpInIndex = mFrameCount; 6168 mBytesRead = 0; 6169 if (mResampler != NULL) { 6170 mResampler->reset(); 6171 } 6172 mActiveTrack->mState = TrackBase::RESUMING; 6173 // signal thread to start 6174 ALOGV("Signal record thread"); 6175 mWaitWorkCV.signal(); 6176 // do not wait for mStartStopCond if exiting 6177 if (exitPending()) { 6178 mActiveTrack.clear(); 6179 status = INVALID_OPERATION; 6180 goto startError; 6181 } 6182 mStartStopCond.wait(mLock); 6183 if (mActiveTrack == 0) { 6184 ALOGV("Record failed to start"); 6185 status = BAD_VALUE; 6186 goto startError; 6187 } 6188 ALOGV("Record started OK"); 6189 return status; 6190 } 6191startError: 6192 AudioSystem::stopInput(mId); 6193 clearSyncStartEvent(); 6194 return status; 6195} 6196 6197void AudioFlinger::RecordThread::clearSyncStartEvent() 6198{ 6199 if (mSyncStartEvent != 0) { 6200 mSyncStartEvent->cancel(); 6201 } 6202 mSyncStartEvent.clear(); 6203 mFramestoDrop = 0; 6204} 6205 6206void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6207{ 6208 sp<SyncEvent> strongEvent = event.promote(); 6209 6210 if (strongEvent != 0) { 6211 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6212 me->handleSyncStartEvent(strongEvent); 6213 } 6214} 6215 6216void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6217{ 6218 if (event == mSyncStartEvent) { 6219 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6220 // from audio HAL 6221 mFramestoDrop = mFrameCount * 2; 6222 } 6223} 6224 6225void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6226 ALOGV("RecordThread::stop"); 6227 sp<ThreadBase> strongMe = this; 6228 { 6229 AutoMutex lock(mLock); 6230 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6231 mActiveTrack->mState = TrackBase::PAUSING; 6232 // do not wait for mStartStopCond if exiting 6233 if (exitPending()) { 6234 return; 6235 } 6236 mStartStopCond.wait(mLock); 6237 // if we have been restarted, recordTrack == mActiveTrack.get() here 6238 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6239 mLock.unlock(); 6240 AudioSystem::stopInput(mId); 6241 mLock.lock(); 6242 ALOGV("Record stopped OK"); 6243 } 6244 } 6245 } 6246} 6247 6248bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6249{ 6250 return false; 6251} 6252 6253status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6254{ 6255 if (!isValidSyncEvent(event)) { 6256 return BAD_VALUE; 6257 } 6258 6259 Mutex::Autolock _l(mLock); 6260 6261 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6262 mTrack->setSyncEvent(event); 6263 return NO_ERROR; 6264 } 6265 return NAME_NOT_FOUND; 6266} 6267 6268status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6269{ 6270 const size_t SIZE = 256; 6271 char buffer[SIZE]; 6272 String8 result; 6273 6274 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6275 result.append(buffer); 6276 6277 if (mActiveTrack != 0) { 6278 result.append("Active Track:\n"); 6279 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6280 mActiveTrack->dump(buffer, SIZE); 6281 result.append(buffer); 6282 6283 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6284 result.append(buffer); 6285 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6286 result.append(buffer); 6287 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6288 result.append(buffer); 6289 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6290 result.append(buffer); 6291 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6292 result.append(buffer); 6293 6294 6295 } else { 6296 result.append("No record client\n"); 6297 } 6298 write(fd, result.string(), result.size()); 6299 6300 dumpBase(fd, args); 6301 dumpEffectChains(fd, args); 6302 6303 return NO_ERROR; 6304} 6305 6306// AudioBufferProvider interface 6307status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6308{ 6309 size_t framesReq = buffer->frameCount; 6310 size_t framesReady = mFrameCount - mRsmpInIndex; 6311 int channelCount; 6312 6313 if (framesReady == 0) { 6314 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6315 if (mBytesRead < 0) { 6316 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6317 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6318 // Force input into standby so that it tries to 6319 // recover at next read attempt 6320 mInput->stream->common.standby(&mInput->stream->common); 6321 usleep(kRecordThreadSleepUs); 6322 } 6323 buffer->raw = NULL; 6324 buffer->frameCount = 0; 6325 return NOT_ENOUGH_DATA; 6326 } 6327 mRsmpInIndex = 0; 6328 framesReady = mFrameCount; 6329 } 6330 6331 if (framesReq > framesReady) { 6332 framesReq = framesReady; 6333 } 6334 6335 if (mChannelCount == 1 && mReqChannelCount == 2) { 6336 channelCount = 1; 6337 } else { 6338 channelCount = 2; 6339 } 6340 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6341 buffer->frameCount = framesReq; 6342 return NO_ERROR; 6343} 6344 6345// AudioBufferProvider interface 6346void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6347{ 6348 mRsmpInIndex += buffer->frameCount; 6349 buffer->frameCount = 0; 6350} 6351 6352bool AudioFlinger::RecordThread::checkForNewParameters_l() 6353{ 6354 bool reconfig = false; 6355 6356 while (!mNewParameters.isEmpty()) { 6357 status_t status = NO_ERROR; 6358 String8 keyValuePair = mNewParameters[0]; 6359 AudioParameter param = AudioParameter(keyValuePair); 6360 int value; 6361 audio_format_t reqFormat = mFormat; 6362 int reqSamplingRate = mReqSampleRate; 6363 int reqChannelCount = mReqChannelCount; 6364 6365 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6366 reqSamplingRate = value; 6367 reconfig = true; 6368 } 6369 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6370 reqFormat = (audio_format_t) value; 6371 reconfig = true; 6372 } 6373 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6374 reqChannelCount = popcount(value); 6375 reconfig = true; 6376 } 6377 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6378 // do not accept frame count changes if tracks are open as the track buffer 6379 // size depends on frame count and correct behavior would not be guaranteed 6380 // if frame count is changed after track creation 6381 if (mActiveTrack != 0) { 6382 status = INVALID_OPERATION; 6383 } else { 6384 reconfig = true; 6385 } 6386 } 6387 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6388 // forward device change to effects that have requested to be 6389 // aware of attached audio device. 6390 for (size_t i = 0; i < mEffectChains.size(); i++) { 6391 mEffectChains[i]->setDevice_l(value); 6392 } 6393 // store input device and output device but do not forward output device to audio HAL. 6394 // Note that status is ignored by the caller for output device 6395 // (see AudioFlinger::setParameters() 6396 if (value & AUDIO_DEVICE_OUT_ALL) { 6397 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6398 status = BAD_VALUE; 6399 } else { 6400 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6401 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6402 if (mTrack != NULL) { 6403 bool suspend = audio_is_bluetooth_sco_device( 6404 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6405 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6406 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6407 } 6408 } 6409 mDevice |= (uint32_t)value; 6410 } 6411 if (status == NO_ERROR) { 6412 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6413 if (status == INVALID_OPERATION) { 6414 mInput->stream->common.standby(&mInput->stream->common); 6415 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6416 keyValuePair.string()); 6417 } 6418 if (reconfig) { 6419 if (status == BAD_VALUE && 6420 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6421 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6422 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6423 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6424 (reqChannelCount <= FCC_2)) { 6425 status = NO_ERROR; 6426 } 6427 if (status == NO_ERROR) { 6428 readInputParameters(); 6429 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6430 } 6431 } 6432 } 6433 6434 mNewParameters.removeAt(0); 6435 6436 mParamStatus = status; 6437 mParamCond.signal(); 6438 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6439 // already timed out waiting for the status and will never signal the condition. 6440 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6441 } 6442 return reconfig; 6443} 6444 6445String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6446{ 6447 char *s; 6448 String8 out_s8 = String8(); 6449 6450 Mutex::Autolock _l(mLock); 6451 if (initCheck() != NO_ERROR) { 6452 return out_s8; 6453 } 6454 6455 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6456 out_s8 = String8(s); 6457 free(s); 6458 return out_s8; 6459} 6460 6461void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6462 AudioSystem::OutputDescriptor desc; 6463 void *param2 = NULL; 6464 6465 switch (event) { 6466 case AudioSystem::INPUT_OPENED: 6467 case AudioSystem::INPUT_CONFIG_CHANGED: 6468 desc.channels = mChannelMask; 6469 desc.samplingRate = mSampleRate; 6470 desc.format = mFormat; 6471 desc.frameCount = mFrameCount; 6472 desc.latency = 0; 6473 param2 = &desc; 6474 break; 6475 6476 case AudioSystem::INPUT_CLOSED: 6477 default: 6478 break; 6479 } 6480 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6481} 6482 6483void AudioFlinger::RecordThread::readInputParameters() 6484{ 6485 delete mRsmpInBuffer; 6486 // mRsmpInBuffer is always assigned a new[] below 6487 delete mRsmpOutBuffer; 6488 mRsmpOutBuffer = NULL; 6489 delete mResampler; 6490 mResampler = NULL; 6491 6492 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6493 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6494 mChannelCount = (uint16_t)popcount(mChannelMask); 6495 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6496 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6497 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6498 mFrameCount = mInputBytes / mFrameSize; 6499 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6500 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6501 6502 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6503 { 6504 int channelCount; 6505 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6506 // stereo to mono post process as the resampler always outputs stereo. 6507 if (mChannelCount == 1 && mReqChannelCount == 2) { 6508 channelCount = 1; 6509 } else { 6510 channelCount = 2; 6511 } 6512 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6513 mResampler->setSampleRate(mSampleRate); 6514 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6515 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6516 6517 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6518 if (mChannelCount == 1 && mReqChannelCount == 1) { 6519 mFrameCount >>= 1; 6520 } 6521 6522 } 6523 mRsmpInIndex = mFrameCount; 6524} 6525 6526unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6527{ 6528 Mutex::Autolock _l(mLock); 6529 if (initCheck() != NO_ERROR) { 6530 return 0; 6531 } 6532 6533 return mInput->stream->get_input_frames_lost(mInput->stream); 6534} 6535 6536uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6537{ 6538 Mutex::Autolock _l(mLock); 6539 uint32_t result = 0; 6540 if (getEffectChain_l(sessionId) != 0) { 6541 result = EFFECT_SESSION; 6542 } 6543 6544 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6545 result |= TRACK_SESSION; 6546 } 6547 6548 return result; 6549} 6550 6551AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6552{ 6553 Mutex::Autolock _l(mLock); 6554 return mTrack; 6555} 6556 6557AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6558{ 6559 Mutex::Autolock _l(mLock); 6560 return mInput; 6561} 6562 6563AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6564{ 6565 Mutex::Autolock _l(mLock); 6566 AudioStreamIn *input = mInput; 6567 mInput = NULL; 6568 return input; 6569} 6570 6571// this method must always be called either with ThreadBase mLock held or inside the thread loop 6572audio_stream_t* AudioFlinger::RecordThread::stream() const 6573{ 6574 if (mInput == NULL) { 6575 return NULL; 6576 } 6577 return &mInput->stream->common; 6578} 6579 6580 6581// ---------------------------------------------------------------------------- 6582 6583audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6584{ 6585 if (!settingsAllowed()) { 6586 return 0; 6587 } 6588 Mutex::Autolock _l(mLock); 6589 return loadHwModule_l(name); 6590} 6591 6592// loadHwModule_l() must be called with AudioFlinger::mLock held 6593audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6594{ 6595 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6596 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6597 ALOGW("loadHwModule() module %s already loaded", name); 6598 return mAudioHwDevs.keyAt(i); 6599 } 6600 } 6601 6602 audio_hw_device_t *dev; 6603 6604 int rc = load_audio_interface(name, &dev); 6605 if (rc) { 6606 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6607 return 0; 6608 } 6609 6610 mHardwareStatus = AUDIO_HW_INIT; 6611 rc = dev->init_check(dev); 6612 mHardwareStatus = AUDIO_HW_IDLE; 6613 if (rc) { 6614 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6615 return 0; 6616 } 6617 6618 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6619 (NULL != dev->set_master_volume)) { 6620 AutoMutex lock(mHardwareLock); 6621 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6622 dev->set_master_volume(dev, mMasterVolume); 6623 mHardwareStatus = AUDIO_HW_IDLE; 6624 } 6625 6626 audio_module_handle_t handle = nextUniqueId(); 6627 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6628 6629 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6630 name, dev->common.module->name, dev->common.module->id, handle); 6631 6632 return handle; 6633 6634} 6635 6636audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6637 audio_devices_t *pDevices, 6638 uint32_t *pSamplingRate, 6639 audio_format_t *pFormat, 6640 audio_channel_mask_t *pChannelMask, 6641 uint32_t *pLatencyMs, 6642 audio_output_flags_t flags) 6643{ 6644 status_t status; 6645 PlaybackThread *thread = NULL; 6646 struct audio_config config = { 6647 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6648 channel_mask: pChannelMask ? *pChannelMask : 0, 6649 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6650 }; 6651 audio_stream_out_t *outStream = NULL; 6652 audio_hw_device_t *outHwDev; 6653 6654 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6655 module, 6656 (pDevices != NULL) ? (int)*pDevices : 0, 6657 config.sample_rate, 6658 config.format, 6659 config.channel_mask, 6660 flags); 6661 6662 if (pDevices == NULL || *pDevices == 0) { 6663 return 0; 6664 } 6665 6666 Mutex::Autolock _l(mLock); 6667 6668 outHwDev = findSuitableHwDev_l(module, *pDevices); 6669 if (outHwDev == NULL) 6670 return 0; 6671 6672 audio_io_handle_t id = nextUniqueId(); 6673 6674 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6675 6676 status = outHwDev->open_output_stream(outHwDev, 6677 id, 6678 *pDevices, 6679 (audio_output_flags_t)flags, 6680 &config, 6681 &outStream); 6682 6683 mHardwareStatus = AUDIO_HW_IDLE; 6684 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6685 outStream, 6686 config.sample_rate, 6687 config.format, 6688 config.channel_mask, 6689 status); 6690 6691 if (status == NO_ERROR && outStream != NULL) { 6692 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6693 6694 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6695 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6696 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6697 thread = new DirectOutputThread(this, output, id, *pDevices); 6698 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6699 } else { 6700 thread = new MixerThread(this, output, id, *pDevices); 6701 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6702 } 6703 mPlaybackThreads.add(id, thread); 6704 6705 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6706 if (pFormat != NULL) *pFormat = config.format; 6707 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6708 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6709 6710 // notify client processes of the new output creation 6711 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6712 6713 // the first primary output opened designates the primary hw device 6714 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6715 ALOGI("Using module %d has the primary audio interface", module); 6716 mPrimaryHardwareDev = outHwDev; 6717 6718 AutoMutex lock(mHardwareLock); 6719 mHardwareStatus = AUDIO_HW_SET_MODE; 6720 outHwDev->set_mode(outHwDev, mMode); 6721 6722 // Determine the level of master volume support the primary audio HAL has, 6723 // and set the initial master volume at the same time. 6724 float initialVolume = 1.0; 6725 mMasterVolumeSupportLvl = MVS_NONE; 6726 6727 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6728 if ((NULL != outHwDev->get_master_volume) && 6729 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6730 mMasterVolumeSupportLvl = MVS_FULL; 6731 } else { 6732 mMasterVolumeSupportLvl = MVS_SETONLY; 6733 initialVolume = 1.0; 6734 } 6735 6736 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6737 if ((NULL == outHwDev->set_master_volume) || 6738 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6739 mMasterVolumeSupportLvl = MVS_NONE; 6740 } 6741 // now that we have a primary device, initialize master volume on other devices 6742 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6743 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6744 6745 if ((dev != mPrimaryHardwareDev) && 6746 (NULL != dev->set_master_volume)) { 6747 dev->set_master_volume(dev, initialVolume); 6748 } 6749 } 6750 mHardwareStatus = AUDIO_HW_IDLE; 6751 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6752 ? initialVolume 6753 : 1.0; 6754 mMasterVolume = initialVolume; 6755 } 6756 return id; 6757 } 6758 6759 return 0; 6760} 6761 6762audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6763 audio_io_handle_t output2) 6764{ 6765 Mutex::Autolock _l(mLock); 6766 MixerThread *thread1 = checkMixerThread_l(output1); 6767 MixerThread *thread2 = checkMixerThread_l(output2); 6768 6769 if (thread1 == NULL || thread2 == NULL) { 6770 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6771 return 0; 6772 } 6773 6774 audio_io_handle_t id = nextUniqueId(); 6775 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6776 thread->addOutputTrack(thread2); 6777 mPlaybackThreads.add(id, thread); 6778 // notify client processes of the new output creation 6779 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6780 return id; 6781} 6782 6783status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6784{ 6785 // keep strong reference on the playback thread so that 6786 // it is not destroyed while exit() is executed 6787 sp<PlaybackThread> thread; 6788 { 6789 Mutex::Autolock _l(mLock); 6790 thread = checkPlaybackThread_l(output); 6791 if (thread == NULL) { 6792 return BAD_VALUE; 6793 } 6794 6795 ALOGV("closeOutput() %d", output); 6796 6797 if (thread->type() == ThreadBase::MIXER) { 6798 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6799 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6800 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6801 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6802 } 6803 } 6804 } 6805 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6806 mPlaybackThreads.removeItem(output); 6807 } 6808 thread->exit(); 6809 // The thread entity (active unit of execution) is no longer running here, 6810 // but the ThreadBase container still exists. 6811 6812 if (thread->type() != ThreadBase::DUPLICATING) { 6813 AudioStreamOut *out = thread->clearOutput(); 6814 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6815 // from now on thread->mOutput is NULL 6816 out->hwDev->close_output_stream(out->hwDev, out->stream); 6817 delete out; 6818 } 6819 return NO_ERROR; 6820} 6821 6822status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6823{ 6824 Mutex::Autolock _l(mLock); 6825 PlaybackThread *thread = checkPlaybackThread_l(output); 6826 6827 if (thread == NULL) { 6828 return BAD_VALUE; 6829 } 6830 6831 ALOGV("suspendOutput() %d", output); 6832 thread->suspend(); 6833 6834 return NO_ERROR; 6835} 6836 6837status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6838{ 6839 Mutex::Autolock _l(mLock); 6840 PlaybackThread *thread = checkPlaybackThread_l(output); 6841 6842 if (thread == NULL) { 6843 return BAD_VALUE; 6844 } 6845 6846 ALOGV("restoreOutput() %d", output); 6847 6848 thread->restore(); 6849 6850 return NO_ERROR; 6851} 6852 6853audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6854 audio_devices_t *pDevices, 6855 uint32_t *pSamplingRate, 6856 audio_format_t *pFormat, 6857 uint32_t *pChannelMask) 6858{ 6859 status_t status; 6860 RecordThread *thread = NULL; 6861 struct audio_config config = { 6862 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6863 channel_mask: pChannelMask ? *pChannelMask : 0, 6864 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6865 }; 6866 uint32_t reqSamplingRate = config.sample_rate; 6867 audio_format_t reqFormat = config.format; 6868 audio_channel_mask_t reqChannels = config.channel_mask; 6869 audio_stream_in_t *inStream = NULL; 6870 audio_hw_device_t *inHwDev; 6871 6872 if (pDevices == NULL || *pDevices == 0) { 6873 return 0; 6874 } 6875 6876 Mutex::Autolock _l(mLock); 6877 6878 inHwDev = findSuitableHwDev_l(module, *pDevices); 6879 if (inHwDev == NULL) 6880 return 0; 6881 6882 audio_io_handle_t id = nextUniqueId(); 6883 6884 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6885 &inStream); 6886 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6887 inStream, 6888 config.sample_rate, 6889 config.format, 6890 config.channel_mask, 6891 status); 6892 6893 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6894 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6895 // or stereo to mono conversions on 16 bit PCM inputs. 6896 if (status == BAD_VALUE && 6897 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6898 (config.sample_rate <= 2 * reqSamplingRate) && 6899 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6900 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6901 inStream = NULL; 6902 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6903 } 6904 6905 if (status == NO_ERROR && inStream != NULL) { 6906 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6907 6908 // Start record thread 6909 // RecorThread require both input and output device indication to forward to audio 6910 // pre processing modules 6911 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6912 thread = new RecordThread(this, 6913 input, 6914 reqSamplingRate, 6915 reqChannels, 6916 id, 6917 device); 6918 mRecordThreads.add(id, thread); 6919 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6920 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6921 if (pFormat != NULL) *pFormat = config.format; 6922 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6923 6924 input->stream->common.standby(&input->stream->common); 6925 6926 // notify client processes of the new input creation 6927 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6928 return id; 6929 } 6930 6931 return 0; 6932} 6933 6934status_t AudioFlinger::closeInput(audio_io_handle_t input) 6935{ 6936 // keep strong reference on the record thread so that 6937 // it is not destroyed while exit() is executed 6938 sp<RecordThread> thread; 6939 { 6940 Mutex::Autolock _l(mLock); 6941 thread = checkRecordThread_l(input); 6942 if (thread == NULL) { 6943 return BAD_VALUE; 6944 } 6945 6946 ALOGV("closeInput() %d", input); 6947 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 6948 mRecordThreads.removeItem(input); 6949 } 6950 thread->exit(); 6951 // The thread entity (active unit of execution) is no longer running here, 6952 // but the ThreadBase container still exists. 6953 6954 AudioStreamIn *in = thread->clearInput(); 6955 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 6956 // from now on thread->mInput is NULL 6957 in->hwDev->close_input_stream(in->hwDev, in->stream); 6958 delete in; 6959 6960 return NO_ERROR; 6961} 6962 6963status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 6964{ 6965 Mutex::Autolock _l(mLock); 6966 MixerThread *dstThread = checkMixerThread_l(output); 6967 if (dstThread == NULL) { 6968 ALOGW("setStreamOutput() bad output id %d", output); 6969 return BAD_VALUE; 6970 } 6971 6972 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 6973 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 6974 6975 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6976 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6977 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 6978 MixerThread *srcThread = (MixerThread *)thread; 6979 srcThread->invalidateTracks(stream); 6980 } 6981 } 6982 6983 return NO_ERROR; 6984} 6985 6986 6987int AudioFlinger::newAudioSessionId() 6988{ 6989 return nextUniqueId(); 6990} 6991 6992void AudioFlinger::acquireAudioSessionId(int audioSession) 6993{ 6994 Mutex::Autolock _l(mLock); 6995 pid_t caller = IPCThreadState::self()->getCallingPid(); 6996 ALOGV("acquiring %d from %d", audioSession, caller); 6997 size_t num = mAudioSessionRefs.size(); 6998 for (size_t i = 0; i< num; i++) { 6999 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7000 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7001 ref->mCnt++; 7002 ALOGV(" incremented refcount to %d", ref->mCnt); 7003 return; 7004 } 7005 } 7006 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7007 ALOGV(" added new entry for %d", audioSession); 7008} 7009 7010void AudioFlinger::releaseAudioSessionId(int audioSession) 7011{ 7012 Mutex::Autolock _l(mLock); 7013 pid_t caller = IPCThreadState::self()->getCallingPid(); 7014 ALOGV("releasing %d from %d", audioSession, caller); 7015 size_t num = mAudioSessionRefs.size(); 7016 for (size_t i = 0; i< num; i++) { 7017 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7018 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7019 ref->mCnt--; 7020 ALOGV(" decremented refcount to %d", ref->mCnt); 7021 if (ref->mCnt == 0) { 7022 mAudioSessionRefs.removeAt(i); 7023 delete ref; 7024 purgeStaleEffects_l(); 7025 } 7026 return; 7027 } 7028 } 7029 ALOGW("session id %d not found for pid %d", audioSession, caller); 7030} 7031 7032void AudioFlinger::purgeStaleEffects_l() { 7033 7034 ALOGV("purging stale effects"); 7035 7036 Vector< sp<EffectChain> > chains; 7037 7038 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7039 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7040 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7041 sp<EffectChain> ec = t->mEffectChains[j]; 7042 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7043 chains.push(ec); 7044 } 7045 } 7046 } 7047 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7048 sp<RecordThread> t = mRecordThreads.valueAt(i); 7049 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7050 sp<EffectChain> ec = t->mEffectChains[j]; 7051 chains.push(ec); 7052 } 7053 } 7054 7055 for (size_t i = 0; i < chains.size(); i++) { 7056 sp<EffectChain> ec = chains[i]; 7057 int sessionid = ec->sessionId(); 7058 sp<ThreadBase> t = ec->mThread.promote(); 7059 if (t == 0) { 7060 continue; 7061 } 7062 size_t numsessionrefs = mAudioSessionRefs.size(); 7063 bool found = false; 7064 for (size_t k = 0; k < numsessionrefs; k++) { 7065 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7066 if (ref->mSessionid == sessionid) { 7067 ALOGV(" session %d still exists for %d with %d refs", 7068 sessionid, ref->mPid, ref->mCnt); 7069 found = true; 7070 break; 7071 } 7072 } 7073 if (!found) { 7074 // remove all effects from the chain 7075 while (ec->mEffects.size()) { 7076 sp<EffectModule> effect = ec->mEffects[0]; 7077 effect->unPin(); 7078 Mutex::Autolock _l (t->mLock); 7079 t->removeEffect_l(effect); 7080 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7081 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7082 if (handle != 0) { 7083 handle->mEffect.clear(); 7084 if (handle->mHasControl && handle->mEnabled) { 7085 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7086 } 7087 } 7088 } 7089 AudioSystem::unregisterEffect(effect->id()); 7090 } 7091 } 7092 } 7093 return; 7094} 7095 7096// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7097AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7098{ 7099 return mPlaybackThreads.valueFor(output).get(); 7100} 7101 7102// checkMixerThread_l() must be called with AudioFlinger::mLock held 7103AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7104{ 7105 PlaybackThread *thread = checkPlaybackThread_l(output); 7106 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7107} 7108 7109// checkRecordThread_l() must be called with AudioFlinger::mLock held 7110AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7111{ 7112 return mRecordThreads.valueFor(input).get(); 7113} 7114 7115uint32_t AudioFlinger::nextUniqueId() 7116{ 7117 return android_atomic_inc(&mNextUniqueId); 7118} 7119 7120AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7121{ 7122 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7123 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7124 AudioStreamOut *output = thread->getOutput(); 7125 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7126 return thread; 7127 } 7128 } 7129 return NULL; 7130} 7131 7132uint32_t AudioFlinger::primaryOutputDevice_l() const 7133{ 7134 PlaybackThread *thread = primaryPlaybackThread_l(); 7135 7136 if (thread == NULL) { 7137 return 0; 7138 } 7139 7140 return thread->device(); 7141} 7142 7143sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7144 int triggerSession, 7145 int listenerSession, 7146 sync_event_callback_t callBack, 7147 void *cookie) 7148{ 7149 Mutex::Autolock _l(mLock); 7150 7151 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7152 status_t playStatus = NAME_NOT_FOUND; 7153 status_t recStatus = NAME_NOT_FOUND; 7154 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7155 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7156 if (playStatus == NO_ERROR) { 7157 return event; 7158 } 7159 } 7160 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7161 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7162 if (recStatus == NO_ERROR) { 7163 return event; 7164 } 7165 } 7166 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7167 mPendingSyncEvents.add(event); 7168 } else { 7169 ALOGV("createSyncEvent() invalid event %d", event->type()); 7170 event.clear(); 7171 } 7172 return event; 7173} 7174 7175// ---------------------------------------------------------------------------- 7176// Effect management 7177// ---------------------------------------------------------------------------- 7178 7179 7180status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7181{ 7182 Mutex::Autolock _l(mLock); 7183 return EffectQueryNumberEffects(numEffects); 7184} 7185 7186status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7187{ 7188 Mutex::Autolock _l(mLock); 7189 return EffectQueryEffect(index, descriptor); 7190} 7191 7192status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7193 effect_descriptor_t *descriptor) const 7194{ 7195 Mutex::Autolock _l(mLock); 7196 return EffectGetDescriptor(pUuid, descriptor); 7197} 7198 7199 7200sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7201 effect_descriptor_t *pDesc, 7202 const sp<IEffectClient>& effectClient, 7203 int32_t priority, 7204 audio_io_handle_t io, 7205 int sessionId, 7206 status_t *status, 7207 int *id, 7208 int *enabled) 7209{ 7210 status_t lStatus = NO_ERROR; 7211 sp<EffectHandle> handle; 7212 effect_descriptor_t desc; 7213 7214 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7215 pid, effectClient.get(), priority, sessionId, io); 7216 7217 if (pDesc == NULL) { 7218 lStatus = BAD_VALUE; 7219 goto Exit; 7220 } 7221 7222 // check audio settings permission for global effects 7223 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7224 lStatus = PERMISSION_DENIED; 7225 goto Exit; 7226 } 7227 7228 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7229 // that can only be created by audio policy manager (running in same process) 7230 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7231 lStatus = PERMISSION_DENIED; 7232 goto Exit; 7233 } 7234 7235 if (io == 0) { 7236 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7237 // output must be specified by AudioPolicyManager when using session 7238 // AUDIO_SESSION_OUTPUT_STAGE 7239 lStatus = BAD_VALUE; 7240 goto Exit; 7241 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7242 // if the output returned by getOutputForEffect() is removed before we lock the 7243 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7244 // and we will exit safely 7245 io = AudioSystem::getOutputForEffect(&desc); 7246 } 7247 } 7248 7249 { 7250 Mutex::Autolock _l(mLock); 7251 7252 7253 if (!EffectIsNullUuid(&pDesc->uuid)) { 7254 // if uuid is specified, request effect descriptor 7255 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7256 if (lStatus < 0) { 7257 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7258 goto Exit; 7259 } 7260 } else { 7261 // if uuid is not specified, look for an available implementation 7262 // of the required type in effect factory 7263 if (EffectIsNullUuid(&pDesc->type)) { 7264 ALOGW("createEffect() no effect type"); 7265 lStatus = BAD_VALUE; 7266 goto Exit; 7267 } 7268 uint32_t numEffects = 0; 7269 effect_descriptor_t d; 7270 d.flags = 0; // prevent compiler warning 7271 bool found = false; 7272 7273 lStatus = EffectQueryNumberEffects(&numEffects); 7274 if (lStatus < 0) { 7275 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7276 goto Exit; 7277 } 7278 for (uint32_t i = 0; i < numEffects; i++) { 7279 lStatus = EffectQueryEffect(i, &desc); 7280 if (lStatus < 0) { 7281 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7282 continue; 7283 } 7284 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7285 // If matching type found save effect descriptor. If the session is 7286 // 0 and the effect is not auxiliary, continue enumeration in case 7287 // an auxiliary version of this effect type is available 7288 found = true; 7289 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7290 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7291 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7292 break; 7293 } 7294 } 7295 } 7296 if (!found) { 7297 lStatus = BAD_VALUE; 7298 ALOGW("createEffect() effect not found"); 7299 goto Exit; 7300 } 7301 // For same effect type, chose auxiliary version over insert version if 7302 // connect to output mix (Compliance to OpenSL ES) 7303 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7304 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7305 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7306 } 7307 } 7308 7309 // Do not allow auxiliary effects on a session different from 0 (output mix) 7310 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7311 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7312 lStatus = INVALID_OPERATION; 7313 goto Exit; 7314 } 7315 7316 // check recording permission for visualizer 7317 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7318 !recordingAllowed()) { 7319 lStatus = PERMISSION_DENIED; 7320 goto Exit; 7321 } 7322 7323 // return effect descriptor 7324 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7325 7326 // If output is not specified try to find a matching audio session ID in one of the 7327 // output threads. 7328 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7329 // because of code checking output when entering the function. 7330 // Note: io is never 0 when creating an effect on an input 7331 if (io == 0) { 7332 // look for the thread where the specified audio session is present 7333 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7334 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7335 io = mPlaybackThreads.keyAt(i); 7336 break; 7337 } 7338 } 7339 if (io == 0) { 7340 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7341 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7342 io = mRecordThreads.keyAt(i); 7343 break; 7344 } 7345 } 7346 } 7347 // If no output thread contains the requested session ID, default to 7348 // first output. The effect chain will be moved to the correct output 7349 // thread when a track with the same session ID is created 7350 if (io == 0 && mPlaybackThreads.size()) { 7351 io = mPlaybackThreads.keyAt(0); 7352 } 7353 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7354 } 7355 ThreadBase *thread = checkRecordThread_l(io); 7356 if (thread == NULL) { 7357 thread = checkPlaybackThread_l(io); 7358 if (thread == NULL) { 7359 ALOGE("createEffect() unknown output thread"); 7360 lStatus = BAD_VALUE; 7361 goto Exit; 7362 } 7363 } 7364 7365 sp<Client> client = registerPid_l(pid); 7366 7367 // create effect on selected output thread 7368 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7369 &desc, enabled, &lStatus); 7370 if (handle != 0 && id != NULL) { 7371 *id = handle->id(); 7372 } 7373 } 7374 7375Exit: 7376 if (status != NULL) { 7377 *status = lStatus; 7378 } 7379 return handle; 7380} 7381 7382status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7383 audio_io_handle_t dstOutput) 7384{ 7385 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7386 sessionId, srcOutput, dstOutput); 7387 Mutex::Autolock _l(mLock); 7388 if (srcOutput == dstOutput) { 7389 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7390 return NO_ERROR; 7391 } 7392 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7393 if (srcThread == NULL) { 7394 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7395 return BAD_VALUE; 7396 } 7397 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7398 if (dstThread == NULL) { 7399 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7400 return BAD_VALUE; 7401 } 7402 7403 Mutex::Autolock _dl(dstThread->mLock); 7404 Mutex::Autolock _sl(srcThread->mLock); 7405 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7406 7407 return NO_ERROR; 7408} 7409 7410// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7411status_t AudioFlinger::moveEffectChain_l(int sessionId, 7412 AudioFlinger::PlaybackThread *srcThread, 7413 AudioFlinger::PlaybackThread *dstThread, 7414 bool reRegister) 7415{ 7416 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7417 sessionId, srcThread, dstThread); 7418 7419 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7420 if (chain == 0) { 7421 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7422 sessionId, srcThread); 7423 return INVALID_OPERATION; 7424 } 7425 7426 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7427 // so that a new chain is created with correct parameters when first effect is added. This is 7428 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7429 // removed. 7430 srcThread->removeEffectChain_l(chain); 7431 7432 // transfer all effects one by one so that new effect chain is created on new thread with 7433 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7434 audio_io_handle_t dstOutput = dstThread->id(); 7435 sp<EffectChain> dstChain; 7436 uint32_t strategy = 0; // prevent compiler warning 7437 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7438 while (effect != 0) { 7439 srcThread->removeEffect_l(effect); 7440 dstThread->addEffect_l(effect); 7441 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7442 if (effect->state() == EffectModule::ACTIVE || 7443 effect->state() == EffectModule::STOPPING) { 7444 effect->start(); 7445 } 7446 // if the move request is not received from audio policy manager, the effect must be 7447 // re-registered with the new strategy and output 7448 if (dstChain == 0) { 7449 dstChain = effect->chain().promote(); 7450 if (dstChain == 0) { 7451 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7452 srcThread->addEffect_l(effect); 7453 return NO_INIT; 7454 } 7455 strategy = dstChain->strategy(); 7456 } 7457 if (reRegister) { 7458 AudioSystem::unregisterEffect(effect->id()); 7459 AudioSystem::registerEffect(&effect->desc(), 7460 dstOutput, 7461 strategy, 7462 sessionId, 7463 effect->id()); 7464 } 7465 effect = chain->getEffectFromId_l(0); 7466 } 7467 7468 return NO_ERROR; 7469} 7470 7471 7472// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7473sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7474 const sp<AudioFlinger::Client>& client, 7475 const sp<IEffectClient>& effectClient, 7476 int32_t priority, 7477 int sessionId, 7478 effect_descriptor_t *desc, 7479 int *enabled, 7480 status_t *status 7481 ) 7482{ 7483 sp<EffectModule> effect; 7484 sp<EffectHandle> handle; 7485 status_t lStatus; 7486 sp<EffectChain> chain; 7487 bool chainCreated = false; 7488 bool effectCreated = false; 7489 bool effectRegistered = false; 7490 7491 lStatus = initCheck(); 7492 if (lStatus != NO_ERROR) { 7493 ALOGW("createEffect_l() Audio driver not initialized."); 7494 goto Exit; 7495 } 7496 7497 // Do not allow effects with session ID 0 on direct output or duplicating threads 7498 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7499 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7500 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7501 desc->name, sessionId); 7502 lStatus = BAD_VALUE; 7503 goto Exit; 7504 } 7505 // Only Pre processor effects are allowed on input threads and only on input threads 7506 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7507 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7508 desc->name, desc->flags, mType); 7509 lStatus = BAD_VALUE; 7510 goto Exit; 7511 } 7512 7513 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7514 7515 { // scope for mLock 7516 Mutex::Autolock _l(mLock); 7517 7518 // check for existing effect chain with the requested audio session 7519 chain = getEffectChain_l(sessionId); 7520 if (chain == 0) { 7521 // create a new chain for this session 7522 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7523 chain = new EffectChain(this, sessionId); 7524 addEffectChain_l(chain); 7525 chain->setStrategy(getStrategyForSession_l(sessionId)); 7526 chainCreated = true; 7527 } else { 7528 effect = chain->getEffectFromDesc_l(desc); 7529 } 7530 7531 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7532 7533 if (effect == 0) { 7534 int id = mAudioFlinger->nextUniqueId(); 7535 // Check CPU and memory usage 7536 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7537 if (lStatus != NO_ERROR) { 7538 goto Exit; 7539 } 7540 effectRegistered = true; 7541 // create a new effect module if none present in the chain 7542 effect = new EffectModule(this, chain, desc, id, sessionId); 7543 lStatus = effect->status(); 7544 if (lStatus != NO_ERROR) { 7545 goto Exit; 7546 } 7547 lStatus = chain->addEffect_l(effect); 7548 if (lStatus != NO_ERROR) { 7549 goto Exit; 7550 } 7551 effectCreated = true; 7552 7553 effect->setDevice(mDevice); 7554 effect->setMode(mAudioFlinger->getMode()); 7555 } 7556 // create effect handle and connect it to effect module 7557 handle = new EffectHandle(effect, client, effectClient, priority); 7558 lStatus = effect->addHandle(handle); 7559 if (enabled != NULL) { 7560 *enabled = (int)effect->isEnabled(); 7561 } 7562 } 7563 7564Exit: 7565 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7566 Mutex::Autolock _l(mLock); 7567 if (effectCreated) { 7568 chain->removeEffect_l(effect); 7569 } 7570 if (effectRegistered) { 7571 AudioSystem::unregisterEffect(effect->id()); 7572 } 7573 if (chainCreated) { 7574 removeEffectChain_l(chain); 7575 } 7576 handle.clear(); 7577 } 7578 7579 if (status != NULL) { 7580 *status = lStatus; 7581 } 7582 return handle; 7583} 7584 7585sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7586{ 7587 sp<EffectChain> chain = getEffectChain_l(sessionId); 7588 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7589} 7590 7591// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7592// PlaybackThread::mLock held 7593status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7594{ 7595 // check for existing effect chain with the requested audio session 7596 int sessionId = effect->sessionId(); 7597 sp<EffectChain> chain = getEffectChain_l(sessionId); 7598 bool chainCreated = false; 7599 7600 if (chain == 0) { 7601 // create a new chain for this session 7602 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7603 chain = new EffectChain(this, sessionId); 7604 addEffectChain_l(chain); 7605 chain->setStrategy(getStrategyForSession_l(sessionId)); 7606 chainCreated = true; 7607 } 7608 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7609 7610 if (chain->getEffectFromId_l(effect->id()) != 0) { 7611 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7612 this, effect->desc().name, chain.get()); 7613 return BAD_VALUE; 7614 } 7615 7616 status_t status = chain->addEffect_l(effect); 7617 if (status != NO_ERROR) { 7618 if (chainCreated) { 7619 removeEffectChain_l(chain); 7620 } 7621 return status; 7622 } 7623 7624 effect->setDevice(mDevice); 7625 effect->setMode(mAudioFlinger->getMode()); 7626 return NO_ERROR; 7627} 7628 7629void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7630 7631 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7632 effect_descriptor_t desc = effect->desc(); 7633 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7634 detachAuxEffect_l(effect->id()); 7635 } 7636 7637 sp<EffectChain> chain = effect->chain().promote(); 7638 if (chain != 0) { 7639 // remove effect chain if removing last effect 7640 if (chain->removeEffect_l(effect) == 0) { 7641 removeEffectChain_l(chain); 7642 } 7643 } else { 7644 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7645 } 7646} 7647 7648void AudioFlinger::ThreadBase::lockEffectChains_l( 7649 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7650{ 7651 effectChains = mEffectChains; 7652 for (size_t i = 0; i < mEffectChains.size(); i++) { 7653 mEffectChains[i]->lock(); 7654 } 7655} 7656 7657void AudioFlinger::ThreadBase::unlockEffectChains( 7658 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7659{ 7660 for (size_t i = 0; i < effectChains.size(); i++) { 7661 effectChains[i]->unlock(); 7662 } 7663} 7664 7665sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7666{ 7667 Mutex::Autolock _l(mLock); 7668 return getEffectChain_l(sessionId); 7669} 7670 7671sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7672{ 7673 size_t size = mEffectChains.size(); 7674 for (size_t i = 0; i < size; i++) { 7675 if (mEffectChains[i]->sessionId() == sessionId) { 7676 return mEffectChains[i]; 7677 } 7678 } 7679 return 0; 7680} 7681 7682void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7683{ 7684 Mutex::Autolock _l(mLock); 7685 size_t size = mEffectChains.size(); 7686 for (size_t i = 0; i < size; i++) { 7687 mEffectChains[i]->setMode_l(mode); 7688 } 7689} 7690 7691void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7692 const wp<EffectHandle>& handle, 7693 bool unpinIfLast) { 7694 7695 Mutex::Autolock _l(mLock); 7696 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7697 // delete the effect module if removing last handle on it 7698 if (effect->removeHandle(handle) == 0) { 7699 if (!effect->isPinned() || unpinIfLast) { 7700 removeEffect_l(effect); 7701 AudioSystem::unregisterEffect(effect->id()); 7702 } 7703 } 7704} 7705 7706status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7707{ 7708 int session = chain->sessionId(); 7709 int16_t *buffer = mMixBuffer; 7710 bool ownsBuffer = false; 7711 7712 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7713 if (session > 0) { 7714 // Only one effect chain can be present in direct output thread and it uses 7715 // the mix buffer as input 7716 if (mType != DIRECT) { 7717 size_t numSamples = mNormalFrameCount * mChannelCount; 7718 buffer = new int16_t[numSamples]; 7719 memset(buffer, 0, numSamples * sizeof(int16_t)); 7720 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7721 ownsBuffer = true; 7722 } 7723 7724 // Attach all tracks with same session ID to this chain. 7725 for (size_t i = 0; i < mTracks.size(); ++i) { 7726 sp<Track> track = mTracks[i]; 7727 if (session == track->sessionId()) { 7728 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7729 track->setMainBuffer(buffer); 7730 chain->incTrackCnt(); 7731 } 7732 } 7733 7734 // indicate all active tracks in the chain 7735 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7736 sp<Track> track = mActiveTracks[i].promote(); 7737 if (track == 0) continue; 7738 if (session == track->sessionId()) { 7739 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7740 chain->incActiveTrackCnt(); 7741 } 7742 } 7743 } 7744 7745 chain->setInBuffer(buffer, ownsBuffer); 7746 chain->setOutBuffer(mMixBuffer); 7747 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7748 // chains list in order to be processed last as it contains output stage effects 7749 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7750 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7751 // after track specific effects and before output stage 7752 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7753 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7754 // Effect chain for other sessions are inserted at beginning of effect 7755 // chains list to be processed before output mix effects. Relative order between other 7756 // sessions is not important 7757 size_t size = mEffectChains.size(); 7758 size_t i = 0; 7759 for (i = 0; i < size; i++) { 7760 if (mEffectChains[i]->sessionId() < session) break; 7761 } 7762 mEffectChains.insertAt(chain, i); 7763 checkSuspendOnAddEffectChain_l(chain); 7764 7765 return NO_ERROR; 7766} 7767 7768size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7769{ 7770 int session = chain->sessionId(); 7771 7772 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7773 7774 for (size_t i = 0; i < mEffectChains.size(); i++) { 7775 if (chain == mEffectChains[i]) { 7776 mEffectChains.removeAt(i); 7777 // detach all active tracks from the chain 7778 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7779 sp<Track> track = mActiveTracks[i].promote(); 7780 if (track == 0) continue; 7781 if (session == track->sessionId()) { 7782 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7783 chain.get(), session); 7784 chain->decActiveTrackCnt(); 7785 } 7786 } 7787 7788 // detach all tracks with same session ID from this chain 7789 for (size_t i = 0; i < mTracks.size(); ++i) { 7790 sp<Track> track = mTracks[i]; 7791 if (session == track->sessionId()) { 7792 track->setMainBuffer(mMixBuffer); 7793 chain->decTrackCnt(); 7794 } 7795 } 7796 break; 7797 } 7798 } 7799 return mEffectChains.size(); 7800} 7801 7802status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7803 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7804{ 7805 Mutex::Autolock _l(mLock); 7806 return attachAuxEffect_l(track, EffectId); 7807} 7808 7809status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7810 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7811{ 7812 status_t status = NO_ERROR; 7813 7814 if (EffectId == 0) { 7815 track->setAuxBuffer(0, NULL); 7816 } else { 7817 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7818 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7819 if (effect != 0) { 7820 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7821 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7822 } else { 7823 status = INVALID_OPERATION; 7824 } 7825 } else { 7826 status = BAD_VALUE; 7827 } 7828 } 7829 return status; 7830} 7831 7832void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7833{ 7834 for (size_t i = 0; i < mTracks.size(); ++i) { 7835 sp<Track> track = mTracks[i]; 7836 if (track->auxEffectId() == effectId) { 7837 attachAuxEffect_l(track, 0); 7838 } 7839 } 7840} 7841 7842status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7843{ 7844 // only one chain per input thread 7845 if (mEffectChains.size() != 0) { 7846 return INVALID_OPERATION; 7847 } 7848 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7849 7850 chain->setInBuffer(NULL); 7851 chain->setOutBuffer(NULL); 7852 7853 checkSuspendOnAddEffectChain_l(chain); 7854 7855 mEffectChains.add(chain); 7856 7857 return NO_ERROR; 7858} 7859 7860size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7861{ 7862 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7863 ALOGW_IF(mEffectChains.size() != 1, 7864 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7865 chain.get(), mEffectChains.size(), this); 7866 if (mEffectChains.size() == 1) { 7867 mEffectChains.removeAt(0); 7868 } 7869 return 0; 7870} 7871 7872// ---------------------------------------------------------------------------- 7873// EffectModule implementation 7874// ---------------------------------------------------------------------------- 7875 7876#undef LOG_TAG 7877#define LOG_TAG "AudioFlinger::EffectModule" 7878 7879AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7880 const wp<AudioFlinger::EffectChain>& chain, 7881 effect_descriptor_t *desc, 7882 int id, 7883 int sessionId) 7884 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7885 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7886{ 7887 ALOGV("Constructor %p", this); 7888 int lStatus; 7889 if (thread == NULL) { 7890 return; 7891 } 7892 7893 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7894 7895 // create effect engine from effect factory 7896 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7897 7898 if (mStatus != NO_ERROR) { 7899 return; 7900 } 7901 lStatus = init(); 7902 if (lStatus < 0) { 7903 mStatus = lStatus; 7904 goto Error; 7905 } 7906 7907 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7908 mPinned = true; 7909 } 7910 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7911 return; 7912Error: 7913 EffectRelease(mEffectInterface); 7914 mEffectInterface = NULL; 7915 ALOGV("Constructor Error %d", mStatus); 7916} 7917 7918AudioFlinger::EffectModule::~EffectModule() 7919{ 7920 ALOGV("Destructor %p", this); 7921 if (mEffectInterface != NULL) { 7922 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7923 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7924 sp<ThreadBase> thread = mThread.promote(); 7925 if (thread != 0) { 7926 audio_stream_t *stream = thread->stream(); 7927 if (stream != NULL) { 7928 stream->remove_audio_effect(stream, mEffectInterface); 7929 } 7930 } 7931 } 7932 // release effect engine 7933 EffectRelease(mEffectInterface); 7934 } 7935} 7936 7937status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7938{ 7939 status_t status; 7940 7941 Mutex::Autolock _l(mLock); 7942 int priority = handle->priority(); 7943 size_t size = mHandles.size(); 7944 sp<EffectHandle> h; 7945 size_t i; 7946 for (i = 0; i < size; i++) { 7947 h = mHandles[i].promote(); 7948 if (h == 0) continue; 7949 if (h->priority() <= priority) break; 7950 } 7951 // if inserted in first place, move effect control from previous owner to this handle 7952 if (i == 0) { 7953 bool enabled = false; 7954 if (h != 0) { 7955 enabled = h->enabled(); 7956 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 7957 } 7958 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 7959 status = NO_ERROR; 7960 } else { 7961 status = ALREADY_EXISTS; 7962 } 7963 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 7964 mHandles.insertAt(handle, i); 7965 return status; 7966} 7967 7968size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 7969{ 7970 Mutex::Autolock _l(mLock); 7971 size_t size = mHandles.size(); 7972 size_t i; 7973 for (i = 0; i < size; i++) { 7974 if (mHandles[i] == handle) break; 7975 } 7976 if (i == size) { 7977 return size; 7978 } 7979 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 7980 7981 bool enabled = false; 7982 EffectHandle *hdl = handle.unsafe_get(); 7983 if (hdl != NULL) { 7984 ALOGV("removeHandle() unsafe_get OK"); 7985 enabled = hdl->enabled(); 7986 } 7987 mHandles.removeAt(i); 7988 size = mHandles.size(); 7989 // if removed from first place, move effect control from this handle to next in line 7990 if (i == 0 && size != 0) { 7991 sp<EffectHandle> h = mHandles[0].promote(); 7992 if (h != 0) { 7993 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 7994 } 7995 } 7996 7997 // Prevent calls to process() and other functions on effect interface from now on. 7998 // The effect engine will be released by the destructor when the last strong reference on 7999 // this object is released which can happen after next process is called. 8000 if (size == 0 && !mPinned) { 8001 mState = DESTROYED; 8002 } 8003 8004 return size; 8005} 8006 8007sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 8008{ 8009 Mutex::Autolock _l(mLock); 8010 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 8011} 8012 8013void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 8014{ 8015 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 8016 // keep a strong reference on this EffectModule to avoid calling the 8017 // destructor before we exit 8018 sp<EffectModule> keep(this); 8019 { 8020 sp<ThreadBase> thread = mThread.promote(); 8021 if (thread != 0) { 8022 thread->disconnectEffect(keep, handle, unpinIfLast); 8023 } 8024 } 8025} 8026 8027void AudioFlinger::EffectModule::updateState() { 8028 Mutex::Autolock _l(mLock); 8029 8030 switch (mState) { 8031 case RESTART: 8032 reset_l(); 8033 // FALL THROUGH 8034 8035 case STARTING: 8036 // clear auxiliary effect input buffer for next accumulation 8037 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8038 memset(mConfig.inputCfg.buffer.raw, 8039 0, 8040 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8041 } 8042 start_l(); 8043 mState = ACTIVE; 8044 break; 8045 case STOPPING: 8046 stop_l(); 8047 mDisableWaitCnt = mMaxDisableWaitCnt; 8048 mState = STOPPED; 8049 break; 8050 case STOPPED: 8051 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8052 // turn off sequence. 8053 if (--mDisableWaitCnt == 0) { 8054 reset_l(); 8055 mState = IDLE; 8056 } 8057 break; 8058 default: //IDLE , ACTIVE, DESTROYED 8059 break; 8060 } 8061} 8062 8063void AudioFlinger::EffectModule::process() 8064{ 8065 Mutex::Autolock _l(mLock); 8066 8067 if (mState == DESTROYED || mEffectInterface == NULL || 8068 mConfig.inputCfg.buffer.raw == NULL || 8069 mConfig.outputCfg.buffer.raw == NULL) { 8070 return; 8071 } 8072 8073 if (isProcessEnabled()) { 8074 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8075 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8076 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8077 mConfig.inputCfg.buffer.s32, 8078 mConfig.inputCfg.buffer.frameCount/2); 8079 } 8080 8081 // do the actual processing in the effect engine 8082 int ret = (*mEffectInterface)->process(mEffectInterface, 8083 &mConfig.inputCfg.buffer, 8084 &mConfig.outputCfg.buffer); 8085 8086 // force transition to IDLE state when engine is ready 8087 if (mState == STOPPED && ret == -ENODATA) { 8088 mDisableWaitCnt = 1; 8089 } 8090 8091 // clear auxiliary effect input buffer for next accumulation 8092 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8093 memset(mConfig.inputCfg.buffer.raw, 0, 8094 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8095 } 8096 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8097 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8098 // If an insert effect is idle and input buffer is different from output buffer, 8099 // accumulate input onto output 8100 sp<EffectChain> chain = mChain.promote(); 8101 if (chain != 0 && chain->activeTrackCnt() != 0) { 8102 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8103 int16_t *in = mConfig.inputCfg.buffer.s16; 8104 int16_t *out = mConfig.outputCfg.buffer.s16; 8105 for (size_t i = 0; i < frameCnt; i++) { 8106 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8107 } 8108 } 8109 } 8110} 8111 8112void AudioFlinger::EffectModule::reset_l() 8113{ 8114 if (mEffectInterface == NULL) { 8115 return; 8116 } 8117 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8118} 8119 8120status_t AudioFlinger::EffectModule::configure() 8121{ 8122 uint32_t channels; 8123 if (mEffectInterface == NULL) { 8124 return NO_INIT; 8125 } 8126 8127 sp<ThreadBase> thread = mThread.promote(); 8128 if (thread == 0) { 8129 return DEAD_OBJECT; 8130 } 8131 8132 // TODO: handle configuration of effects replacing track process 8133 if (thread->channelCount() == 1) { 8134 channels = AUDIO_CHANNEL_OUT_MONO; 8135 } else { 8136 channels = AUDIO_CHANNEL_OUT_STEREO; 8137 } 8138 8139 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8140 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8141 } else { 8142 mConfig.inputCfg.channels = channels; 8143 } 8144 mConfig.outputCfg.channels = channels; 8145 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8146 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8147 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8148 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8149 mConfig.inputCfg.bufferProvider.cookie = NULL; 8150 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8151 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8152 mConfig.outputCfg.bufferProvider.cookie = NULL; 8153 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8154 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8155 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8156 // Insert effect: 8157 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8158 // always overwrites output buffer: input buffer == output buffer 8159 // - in other sessions: 8160 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8161 // other effect: overwrites output buffer: input buffer == output buffer 8162 // Auxiliary effect: 8163 // accumulates in output buffer: input buffer != output buffer 8164 // Therefore: accumulate <=> input buffer != output buffer 8165 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8166 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8167 } else { 8168 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8169 } 8170 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8171 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8172 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8173 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8174 8175 ALOGV("configure() %p thread %p buffer %p framecount %d", 8176 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8177 8178 status_t cmdStatus; 8179 uint32_t size = sizeof(int); 8180 status_t status = (*mEffectInterface)->command(mEffectInterface, 8181 EFFECT_CMD_SET_CONFIG, 8182 sizeof(effect_config_t), 8183 &mConfig, 8184 &size, 8185 &cmdStatus); 8186 if (status == 0) { 8187 status = cmdStatus; 8188 } 8189 8190 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8191 (1000 * mConfig.outputCfg.buffer.frameCount); 8192 8193 return status; 8194} 8195 8196status_t AudioFlinger::EffectModule::init() 8197{ 8198 Mutex::Autolock _l(mLock); 8199 if (mEffectInterface == NULL) { 8200 return NO_INIT; 8201 } 8202 status_t cmdStatus; 8203 uint32_t size = sizeof(status_t); 8204 status_t status = (*mEffectInterface)->command(mEffectInterface, 8205 EFFECT_CMD_INIT, 8206 0, 8207 NULL, 8208 &size, 8209 &cmdStatus); 8210 if (status == 0) { 8211 status = cmdStatus; 8212 } 8213 return status; 8214} 8215 8216status_t AudioFlinger::EffectModule::start() 8217{ 8218 Mutex::Autolock _l(mLock); 8219 return start_l(); 8220} 8221 8222status_t AudioFlinger::EffectModule::start_l() 8223{ 8224 if (mEffectInterface == NULL) { 8225 return NO_INIT; 8226 } 8227 status_t cmdStatus; 8228 uint32_t size = sizeof(status_t); 8229 status_t status = (*mEffectInterface)->command(mEffectInterface, 8230 EFFECT_CMD_ENABLE, 8231 0, 8232 NULL, 8233 &size, 8234 &cmdStatus); 8235 if (status == 0) { 8236 status = cmdStatus; 8237 } 8238 if (status == 0 && 8239 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8240 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8241 sp<ThreadBase> thread = mThread.promote(); 8242 if (thread != 0) { 8243 audio_stream_t *stream = thread->stream(); 8244 if (stream != NULL) { 8245 stream->add_audio_effect(stream, mEffectInterface); 8246 } 8247 } 8248 } 8249 return status; 8250} 8251 8252status_t AudioFlinger::EffectModule::stop() 8253{ 8254 Mutex::Autolock _l(mLock); 8255 return stop_l(); 8256} 8257 8258status_t AudioFlinger::EffectModule::stop_l() 8259{ 8260 if (mEffectInterface == NULL) { 8261 return NO_INIT; 8262 } 8263 status_t cmdStatus; 8264 uint32_t size = sizeof(status_t); 8265 status_t status = (*mEffectInterface)->command(mEffectInterface, 8266 EFFECT_CMD_DISABLE, 8267 0, 8268 NULL, 8269 &size, 8270 &cmdStatus); 8271 if (status == 0) { 8272 status = cmdStatus; 8273 } 8274 if (status == 0 && 8275 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8276 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8277 sp<ThreadBase> thread = mThread.promote(); 8278 if (thread != 0) { 8279 audio_stream_t *stream = thread->stream(); 8280 if (stream != NULL) { 8281 stream->remove_audio_effect(stream, mEffectInterface); 8282 } 8283 } 8284 } 8285 return status; 8286} 8287 8288status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8289 uint32_t cmdSize, 8290 void *pCmdData, 8291 uint32_t *replySize, 8292 void *pReplyData) 8293{ 8294 Mutex::Autolock _l(mLock); 8295// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8296 8297 if (mState == DESTROYED || mEffectInterface == NULL) { 8298 return NO_INIT; 8299 } 8300 status_t status = (*mEffectInterface)->command(mEffectInterface, 8301 cmdCode, 8302 cmdSize, 8303 pCmdData, 8304 replySize, 8305 pReplyData); 8306 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8307 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8308 for (size_t i = 1; i < mHandles.size(); i++) { 8309 sp<EffectHandle> h = mHandles[i].promote(); 8310 if (h != 0) { 8311 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8312 } 8313 } 8314 } 8315 return status; 8316} 8317 8318status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8319{ 8320 8321 Mutex::Autolock _l(mLock); 8322 ALOGV("setEnabled %p enabled %d", this, enabled); 8323 8324 if (enabled != isEnabled()) { 8325 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8326 if (enabled && status != NO_ERROR) { 8327 return status; 8328 } 8329 8330 switch (mState) { 8331 // going from disabled to enabled 8332 case IDLE: 8333 mState = STARTING; 8334 break; 8335 case STOPPED: 8336 mState = RESTART; 8337 break; 8338 case STOPPING: 8339 mState = ACTIVE; 8340 break; 8341 8342 // going from enabled to disabled 8343 case RESTART: 8344 mState = STOPPED; 8345 break; 8346 case STARTING: 8347 mState = IDLE; 8348 break; 8349 case ACTIVE: 8350 mState = STOPPING; 8351 break; 8352 case DESTROYED: 8353 return NO_ERROR; // simply ignore as we are being destroyed 8354 } 8355 for (size_t i = 1; i < mHandles.size(); i++) { 8356 sp<EffectHandle> h = mHandles[i].promote(); 8357 if (h != 0) { 8358 h->setEnabled(enabled); 8359 } 8360 } 8361 } 8362 return NO_ERROR; 8363} 8364 8365bool AudioFlinger::EffectModule::isEnabled() const 8366{ 8367 switch (mState) { 8368 case RESTART: 8369 case STARTING: 8370 case ACTIVE: 8371 return true; 8372 case IDLE: 8373 case STOPPING: 8374 case STOPPED: 8375 case DESTROYED: 8376 default: 8377 return false; 8378 } 8379} 8380 8381bool AudioFlinger::EffectModule::isProcessEnabled() const 8382{ 8383 switch (mState) { 8384 case RESTART: 8385 case ACTIVE: 8386 case STOPPING: 8387 case STOPPED: 8388 return true; 8389 case IDLE: 8390 case STARTING: 8391 case DESTROYED: 8392 default: 8393 return false; 8394 } 8395} 8396 8397status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8398{ 8399 Mutex::Autolock _l(mLock); 8400 status_t status = NO_ERROR; 8401 8402 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8403 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8404 if (isProcessEnabled() && 8405 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8406 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8407 status_t cmdStatus; 8408 uint32_t volume[2]; 8409 uint32_t *pVolume = NULL; 8410 uint32_t size = sizeof(volume); 8411 volume[0] = *left; 8412 volume[1] = *right; 8413 if (controller) { 8414 pVolume = volume; 8415 } 8416 status = (*mEffectInterface)->command(mEffectInterface, 8417 EFFECT_CMD_SET_VOLUME, 8418 size, 8419 volume, 8420 &size, 8421 pVolume); 8422 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8423 *left = volume[0]; 8424 *right = volume[1]; 8425 } 8426 } 8427 return status; 8428} 8429 8430status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8431{ 8432 Mutex::Autolock _l(mLock); 8433 status_t status = NO_ERROR; 8434 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8435 // audio pre processing modules on RecordThread can receive both output and 8436 // input device indication in the same call 8437 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8438 if (dev) { 8439 status_t cmdStatus; 8440 uint32_t size = sizeof(status_t); 8441 8442 status = (*mEffectInterface)->command(mEffectInterface, 8443 EFFECT_CMD_SET_DEVICE, 8444 sizeof(uint32_t), 8445 &dev, 8446 &size, 8447 &cmdStatus); 8448 if (status == NO_ERROR) { 8449 status = cmdStatus; 8450 } 8451 } 8452 dev = device & AUDIO_DEVICE_IN_ALL; 8453 if (dev) { 8454 status_t cmdStatus; 8455 uint32_t size = sizeof(status_t); 8456 8457 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8458 EFFECT_CMD_SET_INPUT_DEVICE, 8459 sizeof(uint32_t), 8460 &dev, 8461 &size, 8462 &cmdStatus); 8463 if (status2 == NO_ERROR) { 8464 status2 = cmdStatus; 8465 } 8466 if (status == NO_ERROR) { 8467 status = status2; 8468 } 8469 } 8470 } 8471 return status; 8472} 8473 8474status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8475{ 8476 Mutex::Autolock _l(mLock); 8477 status_t status = NO_ERROR; 8478 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8479 status_t cmdStatus; 8480 uint32_t size = sizeof(status_t); 8481 status = (*mEffectInterface)->command(mEffectInterface, 8482 EFFECT_CMD_SET_AUDIO_MODE, 8483 sizeof(audio_mode_t), 8484 &mode, 8485 &size, 8486 &cmdStatus); 8487 if (status == NO_ERROR) { 8488 status = cmdStatus; 8489 } 8490 } 8491 return status; 8492} 8493 8494void AudioFlinger::EffectModule::setSuspended(bool suspended) 8495{ 8496 Mutex::Autolock _l(mLock); 8497 mSuspended = suspended; 8498} 8499 8500bool AudioFlinger::EffectModule::suspended() const 8501{ 8502 Mutex::Autolock _l(mLock); 8503 return mSuspended; 8504} 8505 8506status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8507{ 8508 const size_t SIZE = 256; 8509 char buffer[SIZE]; 8510 String8 result; 8511 8512 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8513 result.append(buffer); 8514 8515 bool locked = tryLock(mLock); 8516 // failed to lock - AudioFlinger is probably deadlocked 8517 if (!locked) { 8518 result.append("\t\tCould not lock Fx mutex:\n"); 8519 } 8520 8521 result.append("\t\tSession Status State Engine:\n"); 8522 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8523 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8524 result.append(buffer); 8525 8526 result.append("\t\tDescriptor:\n"); 8527 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8528 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8529 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8530 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8531 result.append(buffer); 8532 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8533 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8534 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8535 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8536 result.append(buffer); 8537 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8538 mDescriptor.apiVersion, 8539 mDescriptor.flags); 8540 result.append(buffer); 8541 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8542 mDescriptor.name); 8543 result.append(buffer); 8544 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8545 mDescriptor.implementor); 8546 result.append(buffer); 8547 8548 result.append("\t\t- Input configuration:\n"); 8549 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8550 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8551 (uint32_t)mConfig.inputCfg.buffer.raw, 8552 mConfig.inputCfg.buffer.frameCount, 8553 mConfig.inputCfg.samplingRate, 8554 mConfig.inputCfg.channels, 8555 mConfig.inputCfg.format); 8556 result.append(buffer); 8557 8558 result.append("\t\t- Output configuration:\n"); 8559 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8560 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8561 (uint32_t)mConfig.outputCfg.buffer.raw, 8562 mConfig.outputCfg.buffer.frameCount, 8563 mConfig.outputCfg.samplingRate, 8564 mConfig.outputCfg.channels, 8565 mConfig.outputCfg.format); 8566 result.append(buffer); 8567 8568 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8569 result.append(buffer); 8570 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8571 for (size_t i = 0; i < mHandles.size(); ++i) { 8572 sp<EffectHandle> handle = mHandles[i].promote(); 8573 if (handle != 0) { 8574 handle->dump(buffer, SIZE); 8575 result.append(buffer); 8576 } 8577 } 8578 8579 result.append("\n"); 8580 8581 write(fd, result.string(), result.length()); 8582 8583 if (locked) { 8584 mLock.unlock(); 8585 } 8586 8587 return NO_ERROR; 8588} 8589 8590// ---------------------------------------------------------------------------- 8591// EffectHandle implementation 8592// ---------------------------------------------------------------------------- 8593 8594#undef LOG_TAG 8595#define LOG_TAG "AudioFlinger::EffectHandle" 8596 8597AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8598 const sp<AudioFlinger::Client>& client, 8599 const sp<IEffectClient>& effectClient, 8600 int32_t priority) 8601 : BnEffect(), 8602 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8603 mPriority(priority), mHasControl(false), mEnabled(false) 8604{ 8605 ALOGV("constructor %p", this); 8606 8607 if (client == 0) { 8608 return; 8609 } 8610 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8611 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8612 if (mCblkMemory != 0) { 8613 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8614 8615 if (mCblk != NULL) { 8616 new(mCblk) effect_param_cblk_t(); 8617 mBuffer = (uint8_t *)mCblk + bufOffset; 8618 } 8619 } else { 8620 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8621 return; 8622 } 8623} 8624 8625AudioFlinger::EffectHandle::~EffectHandle() 8626{ 8627 ALOGV("Destructor %p", this); 8628 disconnect(false); 8629 ALOGV("Destructor DONE %p", this); 8630} 8631 8632status_t AudioFlinger::EffectHandle::enable() 8633{ 8634 ALOGV("enable %p", this); 8635 if (!mHasControl) return INVALID_OPERATION; 8636 if (mEffect == 0) return DEAD_OBJECT; 8637 8638 if (mEnabled) { 8639 return NO_ERROR; 8640 } 8641 8642 mEnabled = true; 8643 8644 sp<ThreadBase> thread = mEffect->thread().promote(); 8645 if (thread != 0) { 8646 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8647 } 8648 8649 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8650 if (mEffect->suspended()) { 8651 return NO_ERROR; 8652 } 8653 8654 status_t status = mEffect->setEnabled(true); 8655 if (status != NO_ERROR) { 8656 if (thread != 0) { 8657 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8658 } 8659 mEnabled = false; 8660 } 8661 return status; 8662} 8663 8664status_t AudioFlinger::EffectHandle::disable() 8665{ 8666 ALOGV("disable %p", this); 8667 if (!mHasControl) return INVALID_OPERATION; 8668 if (mEffect == 0) return DEAD_OBJECT; 8669 8670 if (!mEnabled) { 8671 return NO_ERROR; 8672 } 8673 mEnabled = false; 8674 8675 if (mEffect->suspended()) { 8676 return NO_ERROR; 8677 } 8678 8679 status_t status = mEffect->setEnabled(false); 8680 8681 sp<ThreadBase> thread = mEffect->thread().promote(); 8682 if (thread != 0) { 8683 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8684 } 8685 8686 return status; 8687} 8688 8689void AudioFlinger::EffectHandle::disconnect() 8690{ 8691 disconnect(true); 8692} 8693 8694void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8695{ 8696 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8697 if (mEffect == 0) { 8698 return; 8699 } 8700 mEffect->disconnect(this, unpinIfLast); 8701 8702 if (mHasControl && mEnabled) { 8703 sp<ThreadBase> thread = mEffect->thread().promote(); 8704 if (thread != 0) { 8705 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8706 } 8707 } 8708 8709 // release sp on module => module destructor can be called now 8710 mEffect.clear(); 8711 if (mClient != 0) { 8712 if (mCblk != NULL) { 8713 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8714 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8715 } 8716 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8717 // Client destructor must run with AudioFlinger mutex locked 8718 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8719 mClient.clear(); 8720 } 8721} 8722 8723status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8724 uint32_t cmdSize, 8725 void *pCmdData, 8726 uint32_t *replySize, 8727 void *pReplyData) 8728{ 8729// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8730// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8731 8732 // only get parameter command is permitted for applications not controlling the effect 8733 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8734 return INVALID_OPERATION; 8735 } 8736 if (mEffect == 0) return DEAD_OBJECT; 8737 if (mClient == 0) return INVALID_OPERATION; 8738 8739 // handle commands that are not forwarded transparently to effect engine 8740 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8741 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8742 // no risk to block the whole media server process or mixer threads is we are stuck here 8743 Mutex::Autolock _l(mCblk->lock); 8744 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8745 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8746 mCblk->serverIndex = 0; 8747 mCblk->clientIndex = 0; 8748 return BAD_VALUE; 8749 } 8750 status_t status = NO_ERROR; 8751 while (mCblk->serverIndex < mCblk->clientIndex) { 8752 int reply; 8753 uint32_t rsize = sizeof(int); 8754 int *p = (int *)(mBuffer + mCblk->serverIndex); 8755 int size = *p++; 8756 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8757 ALOGW("command(): invalid parameter block size"); 8758 break; 8759 } 8760 effect_param_t *param = (effect_param_t *)p; 8761 if (param->psize == 0 || param->vsize == 0) { 8762 ALOGW("command(): null parameter or value size"); 8763 mCblk->serverIndex += size; 8764 continue; 8765 } 8766 uint32_t psize = sizeof(effect_param_t) + 8767 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8768 param->vsize; 8769 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8770 psize, 8771 p, 8772 &rsize, 8773 &reply); 8774 // stop at first error encountered 8775 if (ret != NO_ERROR) { 8776 status = ret; 8777 *(int *)pReplyData = reply; 8778 break; 8779 } else if (reply != NO_ERROR) { 8780 *(int *)pReplyData = reply; 8781 break; 8782 } 8783 mCblk->serverIndex += size; 8784 } 8785 mCblk->serverIndex = 0; 8786 mCblk->clientIndex = 0; 8787 return status; 8788 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8789 *(int *)pReplyData = NO_ERROR; 8790 return enable(); 8791 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8792 *(int *)pReplyData = NO_ERROR; 8793 return disable(); 8794 } 8795 8796 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8797} 8798 8799void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8800{ 8801 ALOGV("setControl %p control %d", this, hasControl); 8802 8803 mHasControl = hasControl; 8804 mEnabled = enabled; 8805 8806 if (signal && mEffectClient != 0) { 8807 mEffectClient->controlStatusChanged(hasControl); 8808 } 8809} 8810 8811void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8812 uint32_t cmdSize, 8813 void *pCmdData, 8814 uint32_t replySize, 8815 void *pReplyData) 8816{ 8817 if (mEffectClient != 0) { 8818 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8819 } 8820} 8821 8822 8823 8824void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8825{ 8826 if (mEffectClient != 0) { 8827 mEffectClient->enableStatusChanged(enabled); 8828 } 8829} 8830 8831status_t AudioFlinger::EffectHandle::onTransact( 8832 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8833{ 8834 return BnEffect::onTransact(code, data, reply, flags); 8835} 8836 8837 8838void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8839{ 8840 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8841 8842 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8843 (mClient == 0) ? getpid_cached : mClient->pid(), 8844 mPriority, 8845 mHasControl, 8846 !locked, 8847 mCblk ? mCblk->clientIndex : 0, 8848 mCblk ? mCblk->serverIndex : 0 8849 ); 8850 8851 if (locked) { 8852 mCblk->lock.unlock(); 8853 } 8854} 8855 8856#undef LOG_TAG 8857#define LOG_TAG "AudioFlinger::EffectChain" 8858 8859AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8860 int sessionId) 8861 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8862 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8863 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8864{ 8865 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8866 if (thread == NULL) { 8867 return; 8868 } 8869 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8870 thread->frameCount(); 8871} 8872 8873AudioFlinger::EffectChain::~EffectChain() 8874{ 8875 if (mOwnInBuffer) { 8876 delete mInBuffer; 8877 } 8878 8879} 8880 8881// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8882sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8883{ 8884 size_t size = mEffects.size(); 8885 8886 for (size_t i = 0; i < size; i++) { 8887 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8888 return mEffects[i]; 8889 } 8890 } 8891 return 0; 8892} 8893 8894// getEffectFromId_l() must be called with ThreadBase::mLock held 8895sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8896{ 8897 size_t size = mEffects.size(); 8898 8899 for (size_t i = 0; i < size; i++) { 8900 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8901 if (id == 0 || mEffects[i]->id() == id) { 8902 return mEffects[i]; 8903 } 8904 } 8905 return 0; 8906} 8907 8908// getEffectFromType_l() must be called with ThreadBase::mLock held 8909sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8910 const effect_uuid_t *type) 8911{ 8912 size_t size = mEffects.size(); 8913 8914 for (size_t i = 0; i < size; i++) { 8915 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8916 return mEffects[i]; 8917 } 8918 } 8919 return 0; 8920} 8921 8922void AudioFlinger::EffectChain::clearInputBuffer() 8923{ 8924 Mutex::Autolock _l(mLock); 8925 sp<ThreadBase> thread = mThread.promote(); 8926 if (thread == 0) { 8927 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 8928 return; 8929 } 8930 clearInputBuffer_l(thread); 8931} 8932 8933// Must be called with EffectChain::mLock locked 8934void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 8935{ 8936 size_t numSamples = thread->frameCount() * thread->channelCount(); 8937 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 8938 8939} 8940 8941// Must be called with EffectChain::mLock locked 8942void AudioFlinger::EffectChain::process_l() 8943{ 8944 sp<ThreadBase> thread = mThread.promote(); 8945 if (thread == 0) { 8946 ALOGW("process_l(): cannot promote mixer thread"); 8947 return; 8948 } 8949 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 8950 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 8951 // always process effects unless no more tracks are on the session and the effect tail 8952 // has been rendered 8953 bool doProcess = true; 8954 if (!isGlobalSession) { 8955 bool tracksOnSession = (trackCnt() != 0); 8956 8957 if (!tracksOnSession && mTailBufferCount == 0) { 8958 doProcess = false; 8959 } 8960 8961 if (activeTrackCnt() == 0) { 8962 // if no track is active and the effect tail has not been rendered, 8963 // the input buffer must be cleared here as the mixer process will not do it 8964 if (tracksOnSession || mTailBufferCount > 0) { 8965 clearInputBuffer_l(thread); 8966 if (mTailBufferCount > 0) { 8967 mTailBufferCount--; 8968 } 8969 } 8970 } 8971 } 8972 8973 size_t size = mEffects.size(); 8974 if (doProcess) { 8975 for (size_t i = 0; i < size; i++) { 8976 mEffects[i]->process(); 8977 } 8978 } 8979 for (size_t i = 0; i < size; i++) { 8980 mEffects[i]->updateState(); 8981 } 8982} 8983 8984// addEffect_l() must be called with PlaybackThread::mLock held 8985status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 8986{ 8987 effect_descriptor_t desc = effect->desc(); 8988 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 8989 8990 Mutex::Autolock _l(mLock); 8991 effect->setChain(this); 8992 sp<ThreadBase> thread = mThread.promote(); 8993 if (thread == 0) { 8994 return NO_INIT; 8995 } 8996 effect->setThread(thread); 8997 8998 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8999 // Auxiliary effects are inserted at the beginning of mEffects vector as 9000 // they are processed first and accumulated in chain input buffer 9001 mEffects.insertAt(effect, 0); 9002 9003 // the input buffer for auxiliary effect contains mono samples in 9004 // 32 bit format. This is to avoid saturation in AudoMixer 9005 // accumulation stage. Saturation is done in EffectModule::process() before 9006 // calling the process in effect engine 9007 size_t numSamples = thread->frameCount(); 9008 int32_t *buffer = new int32_t[numSamples]; 9009 memset(buffer, 0, numSamples * sizeof(int32_t)); 9010 effect->setInBuffer((int16_t *)buffer); 9011 // auxiliary effects output samples to chain input buffer for further processing 9012 // by insert effects 9013 effect->setOutBuffer(mInBuffer); 9014 } else { 9015 // Insert effects are inserted at the end of mEffects vector as they are processed 9016 // after track and auxiliary effects. 9017 // Insert effect order as a function of indicated preference: 9018 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9019 // another effect is present 9020 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9021 // last effect claiming first position 9022 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9023 // first effect claiming last position 9024 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9025 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9026 // already present 9027 9028 size_t size = mEffects.size(); 9029 size_t idx_insert = size; 9030 ssize_t idx_insert_first = -1; 9031 ssize_t idx_insert_last = -1; 9032 9033 for (size_t i = 0; i < size; i++) { 9034 effect_descriptor_t d = mEffects[i]->desc(); 9035 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9036 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9037 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9038 // check invalid effect chaining combinations 9039 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9040 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9041 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9042 return INVALID_OPERATION; 9043 } 9044 // remember position of first insert effect and by default 9045 // select this as insert position for new effect 9046 if (idx_insert == size) { 9047 idx_insert = i; 9048 } 9049 // remember position of last insert effect claiming 9050 // first position 9051 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9052 idx_insert_first = i; 9053 } 9054 // remember position of first insert effect claiming 9055 // last position 9056 if (iPref == EFFECT_FLAG_INSERT_LAST && 9057 idx_insert_last == -1) { 9058 idx_insert_last = i; 9059 } 9060 } 9061 } 9062 9063 // modify idx_insert from first position if needed 9064 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9065 if (idx_insert_last != -1) { 9066 idx_insert = idx_insert_last; 9067 } else { 9068 idx_insert = size; 9069 } 9070 } else { 9071 if (idx_insert_first != -1) { 9072 idx_insert = idx_insert_first + 1; 9073 } 9074 } 9075 9076 // always read samples from chain input buffer 9077 effect->setInBuffer(mInBuffer); 9078 9079 // if last effect in the chain, output samples to chain 9080 // output buffer, otherwise to chain input buffer 9081 if (idx_insert == size) { 9082 if (idx_insert != 0) { 9083 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9084 mEffects[idx_insert-1]->configure(); 9085 } 9086 effect->setOutBuffer(mOutBuffer); 9087 } else { 9088 effect->setOutBuffer(mInBuffer); 9089 } 9090 mEffects.insertAt(effect, idx_insert); 9091 9092 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9093 } 9094 effect->configure(); 9095 return NO_ERROR; 9096} 9097 9098// removeEffect_l() must be called with PlaybackThread::mLock held 9099size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9100{ 9101 Mutex::Autolock _l(mLock); 9102 size_t size = mEffects.size(); 9103 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9104 9105 for (size_t i = 0; i < size; i++) { 9106 if (effect == mEffects[i]) { 9107 // calling stop here will remove pre-processing effect from the audio HAL. 9108 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9109 // the middle of a read from audio HAL 9110 if (mEffects[i]->state() == EffectModule::ACTIVE || 9111 mEffects[i]->state() == EffectModule::STOPPING) { 9112 mEffects[i]->stop(); 9113 } 9114 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9115 delete[] effect->inBuffer(); 9116 } else { 9117 if (i == size - 1 && i != 0) { 9118 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9119 mEffects[i - 1]->configure(); 9120 } 9121 } 9122 mEffects.removeAt(i); 9123 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9124 break; 9125 } 9126 } 9127 9128 return mEffects.size(); 9129} 9130 9131// setDevice_l() must be called with PlaybackThread::mLock held 9132void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9133{ 9134 size_t size = mEffects.size(); 9135 for (size_t i = 0; i < size; i++) { 9136 mEffects[i]->setDevice(device); 9137 } 9138} 9139 9140// setMode_l() must be called with PlaybackThread::mLock held 9141void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9142{ 9143 size_t size = mEffects.size(); 9144 for (size_t i = 0; i < size; i++) { 9145 mEffects[i]->setMode(mode); 9146 } 9147} 9148 9149// setVolume_l() must be called with PlaybackThread::mLock held 9150bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9151{ 9152 uint32_t newLeft = *left; 9153 uint32_t newRight = *right; 9154 bool hasControl = false; 9155 int ctrlIdx = -1; 9156 size_t size = mEffects.size(); 9157 9158 // first update volume controller 9159 for (size_t i = size; i > 0; i--) { 9160 if (mEffects[i - 1]->isProcessEnabled() && 9161 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9162 ctrlIdx = i - 1; 9163 hasControl = true; 9164 break; 9165 } 9166 } 9167 9168 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9169 if (hasControl) { 9170 *left = mNewLeftVolume; 9171 *right = mNewRightVolume; 9172 } 9173 return hasControl; 9174 } 9175 9176 mVolumeCtrlIdx = ctrlIdx; 9177 mLeftVolume = newLeft; 9178 mRightVolume = newRight; 9179 9180 // second get volume update from volume controller 9181 if (ctrlIdx >= 0) { 9182 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9183 mNewLeftVolume = newLeft; 9184 mNewRightVolume = newRight; 9185 } 9186 // then indicate volume to all other effects in chain. 9187 // Pass altered volume to effects before volume controller 9188 // and requested volume to effects after controller 9189 uint32_t lVol = newLeft; 9190 uint32_t rVol = newRight; 9191 9192 for (size_t i = 0; i < size; i++) { 9193 if ((int)i == ctrlIdx) continue; 9194 // this also works for ctrlIdx == -1 when there is no volume controller 9195 if ((int)i > ctrlIdx) { 9196 lVol = *left; 9197 rVol = *right; 9198 } 9199 mEffects[i]->setVolume(&lVol, &rVol, false); 9200 } 9201 *left = newLeft; 9202 *right = newRight; 9203 9204 return hasControl; 9205} 9206 9207status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9208{ 9209 const size_t SIZE = 256; 9210 char buffer[SIZE]; 9211 String8 result; 9212 9213 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9214 result.append(buffer); 9215 9216 bool locked = tryLock(mLock); 9217 // failed to lock - AudioFlinger is probably deadlocked 9218 if (!locked) { 9219 result.append("\tCould not lock mutex:\n"); 9220 } 9221 9222 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9223 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9224 mEffects.size(), 9225 (uint32_t)mInBuffer, 9226 (uint32_t)mOutBuffer, 9227 mActiveTrackCnt); 9228 result.append(buffer); 9229 write(fd, result.string(), result.size()); 9230 9231 for (size_t i = 0; i < mEffects.size(); ++i) { 9232 sp<EffectModule> effect = mEffects[i]; 9233 if (effect != 0) { 9234 effect->dump(fd, args); 9235 } 9236 } 9237 9238 if (locked) { 9239 mLock.unlock(); 9240 } 9241 9242 return NO_ERROR; 9243} 9244 9245// must be called with ThreadBase::mLock held 9246void AudioFlinger::EffectChain::setEffectSuspended_l( 9247 const effect_uuid_t *type, bool suspend) 9248{ 9249 sp<SuspendedEffectDesc> desc; 9250 // use effect type UUID timelow as key as there is no real risk of identical 9251 // timeLow fields among effect type UUIDs. 9252 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9253 if (suspend) { 9254 if (index >= 0) { 9255 desc = mSuspendedEffects.valueAt(index); 9256 } else { 9257 desc = new SuspendedEffectDesc(); 9258 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9259 mSuspendedEffects.add(type->timeLow, desc); 9260 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9261 } 9262 if (desc->mRefCount++ == 0) { 9263 sp<EffectModule> effect = getEffectIfEnabled(type); 9264 if (effect != 0) { 9265 desc->mEffect = effect; 9266 effect->setSuspended(true); 9267 effect->setEnabled(false); 9268 } 9269 } 9270 } else { 9271 if (index < 0) { 9272 return; 9273 } 9274 desc = mSuspendedEffects.valueAt(index); 9275 if (desc->mRefCount <= 0) { 9276 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9277 desc->mRefCount = 1; 9278 } 9279 if (--desc->mRefCount == 0) { 9280 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9281 if (desc->mEffect != 0) { 9282 sp<EffectModule> effect = desc->mEffect.promote(); 9283 if (effect != 0) { 9284 effect->setSuspended(false); 9285 sp<EffectHandle> handle = effect->controlHandle(); 9286 if (handle != 0) { 9287 effect->setEnabled(handle->enabled()); 9288 } 9289 } 9290 desc->mEffect.clear(); 9291 } 9292 mSuspendedEffects.removeItemsAt(index); 9293 } 9294 } 9295} 9296 9297// must be called with ThreadBase::mLock held 9298void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9299{ 9300 sp<SuspendedEffectDesc> desc; 9301 9302 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9303 if (suspend) { 9304 if (index >= 0) { 9305 desc = mSuspendedEffects.valueAt(index); 9306 } else { 9307 desc = new SuspendedEffectDesc(); 9308 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9309 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9310 } 9311 if (desc->mRefCount++ == 0) { 9312 Vector< sp<EffectModule> > effects; 9313 getSuspendEligibleEffects(effects); 9314 for (size_t i = 0; i < effects.size(); i++) { 9315 setEffectSuspended_l(&effects[i]->desc().type, true); 9316 } 9317 } 9318 } else { 9319 if (index < 0) { 9320 return; 9321 } 9322 desc = mSuspendedEffects.valueAt(index); 9323 if (desc->mRefCount <= 0) { 9324 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9325 desc->mRefCount = 1; 9326 } 9327 if (--desc->mRefCount == 0) { 9328 Vector<const effect_uuid_t *> types; 9329 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9330 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9331 continue; 9332 } 9333 types.add(&mSuspendedEffects.valueAt(i)->mType); 9334 } 9335 for (size_t i = 0; i < types.size(); i++) { 9336 setEffectSuspended_l(types[i], false); 9337 } 9338 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9339 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9340 } 9341 } 9342} 9343 9344 9345// The volume effect is used for automated tests only 9346#ifndef OPENSL_ES_H_ 9347static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9348 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9349const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9350#endif //OPENSL_ES_H_ 9351 9352bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9353{ 9354 // auxiliary effects and visualizer are never suspended on output mix 9355 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9356 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9357 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9358 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9359 return false; 9360 } 9361 return true; 9362} 9363 9364void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9365{ 9366 effects.clear(); 9367 for (size_t i = 0; i < mEffects.size(); i++) { 9368 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9369 effects.add(mEffects[i]); 9370 } 9371 } 9372} 9373 9374sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9375 const effect_uuid_t *type) 9376{ 9377 sp<EffectModule> effect = getEffectFromType_l(type); 9378 return effect != 0 && effect->isEnabled() ? effect : 0; 9379} 9380 9381void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9382 bool enabled) 9383{ 9384 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9385 if (enabled) { 9386 if (index < 0) { 9387 // if the effect is not suspend check if all effects are suspended 9388 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9389 if (index < 0) { 9390 return; 9391 } 9392 if (!isEffectEligibleForSuspend(effect->desc())) { 9393 return; 9394 } 9395 setEffectSuspended_l(&effect->desc().type, enabled); 9396 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9397 if (index < 0) { 9398 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9399 return; 9400 } 9401 } 9402 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9403 effect->desc().type.timeLow); 9404 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9405 // if effect is requested to suspended but was not yet enabled, supend it now. 9406 if (desc->mEffect == 0) { 9407 desc->mEffect = effect; 9408 effect->setEnabled(false); 9409 effect->setSuspended(true); 9410 } 9411 } else { 9412 if (index < 0) { 9413 return; 9414 } 9415 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9416 effect->desc().type.timeLow); 9417 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9418 desc->mEffect.clear(); 9419 effect->setSuspended(false); 9420 } 9421} 9422 9423#undef LOG_TAG 9424#define LOG_TAG "AudioFlinger" 9425 9426// ---------------------------------------------------------------------------- 9427 9428status_t AudioFlinger::onTransact( 9429 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9430{ 9431 return BnAudioFlinger::onTransact(code, data, reply, flags); 9432} 9433 9434}; // namespace android 9435