/services/audioflinger/Threads.cpp
C++ | 6705 lines | 5161 code | 676 blank | 868 comment | 1358 complexity | da688f455a21da8581ee6b05412f94b3 MD5 | raw file
- /*
- ** Copyright (c) 2013, The Linux Foundation. All rights reserved.
- ** Not a Contribution.
- ** Copyright 2012, The Android Open Source Project
- **
- ** Licensed under the Apache License, Version 2.0 (the "License");
- ** you may not use this file except in compliance with the License.
- ** You may obtain a copy of the License at
- **
- ** http://www.apache.org/licenses/LICENSE-2.0
- **
- ** Unless required by applicable law or agreed to in writing, software
- ** distributed under the License is distributed on an "AS IS" BASIS,
- ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- ** See the License for the specific language governing permissions and
- ** limitations under the License.
- */
- #define LOG_TAG "AudioFlinger"
- //#define LOG_NDEBUG 0
- #define ATRACE_TAG ATRACE_TAG_AUDIO
- #include "Configuration.h"
- #include <math.h>
- #include <fcntl.h>
- #include <sys/stat.h>
- #include <cutils/properties.h>
- #include <media/AudioParameter.h>
- #include <media/AudioResamplerPublic.h>
- #include <utils/Log.h>
- #include <utils/Trace.h>
- #include <private/media/AudioTrackShared.h>
- #include <hardware/audio.h>
- #include <audio_effects/effect_ns.h>
- #include <audio_effects/effect_aec.h>
- #include <audio_utils/primitives.h>
- #include <audio_utils/format.h>
- #include <audio_utils/minifloat.h>
- // NBAIO implementations
- #include <media/nbaio/AudioStreamInSource.h>
- #include <media/nbaio/AudioStreamOutSink.h>
- #include <media/nbaio/MonoPipe.h>
- #include <media/nbaio/MonoPipeReader.h>
- #include <media/nbaio/Pipe.h>
- #include <media/nbaio/PipeReader.h>
- #include <media/nbaio/SourceAudioBufferProvider.h>
- #include <powermanager/PowerManager.h>
- #include <common_time/cc_helper.h>
- #include <common_time/local_clock.h>
- #include "AudioFlinger.h"
- #include "AudioMixer.h"
- #include "FastMixer.h"
- #include "FastCapture.h"
- #include "ServiceUtilities.h"
- #include "SchedulingPolicyService.h"
- #ifdef ADD_BATTERY_DATA
- #include <media/IMediaPlayerService.h>
- #include <media/IMediaDeathNotifier.h>
- #endif
- #ifdef DEBUG_CPU_USAGE
- #include <cpustats/CentralTendencyStatistics.h>
- #include <cpustats/ThreadCpuUsage.h>
- #endif
- // ----------------------------------------------------------------------------
- // Note: the following macro is used for extremely verbose logging message. In
- // order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
- // 0; but one side effect of this is to turn all LOGV's as well. Some messages
- // are so verbose that we want to suppress them even when we have ALOG_ASSERT
- // turned on. Do not uncomment the #def below unless you really know what you
- // are doing and want to see all of the extremely verbose messages.
- //#define VERY_VERY_VERBOSE_LOGGING
- #ifdef VERY_VERY_VERBOSE_LOGGING
- #define ALOGVV ALOGV
- #else
- #define ALOGVV(a...) do { } while(0)
- #endif
- #define max(a, b) ((a) > (b) ? (a) : (b))
- #ifdef QCOM_DIRECTTRACK
- #define DIRECT_TRACK_EOS 1
- #define DIRECT_TRACK_HW_FAIL 6
- static const char lockName[] = "DirectTrack";
- #endif
- namespace android {
- // retry counts for buffer fill timeout
- // 50 * ~20msecs = 1 second
- static const int8_t kMaxTrackRetries = 50;
- static const int8_t kMaxTrackStartupRetries = 50;
- // allow less retry attempts on direct output thread.
- // direct outputs can be a scarce resource in audio hardware and should
- // be released as quickly as possible.
- static const int8_t kMaxTrackRetriesDirect = 2;
- // don't warn about blocked writes or record buffer overflows more often than this
- static const nsecs_t kWarningThrottleNs = seconds(5);
- // RecordThread loop sleep time upon application overrun or audio HAL read error
- static const int kRecordThreadSleepUs = 5000;
- // maximum time to wait in sendConfigEvent_l() for a status to be received
- static const nsecs_t kConfigEventTimeoutNs = seconds(2);
- // minimum sleep time for the mixer thread loop when tracks are active but in underrun
- static const uint32_t kMinThreadSleepTimeUs = 5000;
- // maximum divider applied to the active sleep time in the mixer thread loop
- static const uint32_t kMaxThreadSleepTimeShift = 2;
- // minimum normal sink buffer size, expressed in milliseconds rather than frames
- static const uint32_t kMinNormalSinkBufferSizeMs = 20;
- // maximum normal sink buffer size
- static const uint32_t kMaxNormalSinkBufferSizeMs = 24;
- // Offloaded output thread standby delay: allows track transition without going to standby
- static const nsecs_t kOffloadStandbyDelayNs = seconds(1);
- // Whether to use fast mixer
- static const enum {
- FastMixer_Never, // never initialize or use: for debugging only
- FastMixer_Always, // always initialize and use, even if not needed: for debugging only
- // normal mixer multiplier is 1
- FastMixer_Static, // initialize if needed, then use all the time if initialized,
- // multiplier is calculated based on min & max normal mixer buffer size
- FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load,
- // multiplier is calculated based on min & max normal mixer buffer size
- // FIXME for FastMixer_Dynamic:
- // Supporting this option will require fixing HALs that can't handle large writes.
- // For example, one HAL implementation returns an error from a large write,
- // and another HAL implementation corrupts memory, possibly in the sample rate converter.
- // We could either fix the HAL implementations, or provide a wrapper that breaks
- // up large writes into smaller ones, and the wrapper would need to deal with scheduler.
- } kUseFastMixer = FastMixer_Static;
- // Whether to use fast capture
- static const enum {
- FastCapture_Never, // never initialize or use: for debugging only
- FastCapture_Always, // always initialize and use, even if not needed: for debugging only
- FastCapture_Static, // initialize if needed, then use all the time if initialized
- } kUseFastCapture = FastCapture_Static;
- // Priorities for requestPriority
- static const int kPriorityAudioApp = 2;
- static const int kPriorityFastMixer = 3;
- static const int kPriorityFastCapture = 3;
- // IAudioFlinger::createTrack() reports back to client the total size of shared memory area
- // for the track. The client then sub-divides this into smaller buffers for its use.
- // Currently the client uses N-buffering by default, but doesn't tell us about the value of N.
- // So for now we just assume that client is double-buffered for fast tracks.
- // FIXME It would be better for client to tell AudioFlinger the value of N,
- // so AudioFlinger could allocate the right amount of memory.
- // See the client's minBufCount and mNotificationFramesAct calculations for details.
- // This is the default value, if not specified by property.
- static const int kFastTrackMultiplier = 2;
- // The minimum and maximum allowed values
- static const int kFastTrackMultiplierMin = 1;
- static const int kFastTrackMultiplierMax = 2;
- // The actual value to use, which can be specified per-device via property af.fast_track_multiplier.
- static int sFastTrackMultiplier = kFastTrackMultiplier;
- // See Thread::readOnlyHeap().
- // Initially this heap is used to allocate client buffers for "fast" AudioRecord.
- // Eventually it will be the single buffer that FastCapture writes into via HAL read(),
- // and that all "fast" AudioRecord clients read from. In either case, the size can be small.
- static const size_t kRecordThreadReadOnlyHeapSize = 0x2000;
- // ----------------------------------------------------------------------------
- static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT;
- static void sFastTrackMultiplierInit()
- {
- char value[PROPERTY_VALUE_MAX];
- if (property_get("af.fast_track_multiplier", value, NULL) > 0) {
- char *endptr;
- unsigned long ul = strtoul(value, &endptr, 0);
- if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) {
- sFastTrackMultiplier = (int) ul;
- }
- }
- }
- // ----------------------------------------------------------------------------
- #ifdef ADD_BATTERY_DATA
- // To collect the amplifier usage
- static void addBatteryData(uint32_t params) {
- sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
- if (service == NULL) {
- // it already logged
- return;
- }
- service->addBatteryData(params);
- }
- #endif
- // ----------------------------------------------------------------------------
- // CPU Stats
- // ----------------------------------------------------------------------------
- class CpuStats {
- public:
- CpuStats();
- void sample(const String8 &title);
- #ifdef DEBUG_CPU_USAGE
- private:
- ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns
- CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns
- CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles
- int mCpuNum; // thread's current CPU number
- int mCpukHz; // frequency of thread's current CPU in kHz
- #endif
- };
- CpuStats::CpuStats()
- #ifdef DEBUG_CPU_USAGE
- : mCpuNum(-1), mCpukHz(-1)
- #endif
- {
- }
- void CpuStats::sample(const String8 &title
- #ifndef DEBUG_CPU_USAGE
- __unused
- #endif
- ) {
- #ifdef DEBUG_CPU_USAGE
- // get current thread's delta CPU time in wall clock ns
- double wcNs;
- bool valid = mCpuUsage.sampleAndEnable(wcNs);
- // record sample for wall clock statistics
- if (valid) {
- mWcStats.sample(wcNs);
- }
- // get the current CPU number
- int cpuNum = sched_getcpu();
- // get the current CPU frequency in kHz
- int cpukHz = mCpuUsage.getCpukHz(cpuNum);
- // check if either CPU number or frequency changed
- if (cpuNum != mCpuNum || cpukHz != mCpukHz) {
- mCpuNum = cpuNum;
- mCpukHz = cpukHz;
- // ignore sample for purposes of cycles
- valid = false;
- }
- // if no change in CPU number or frequency, then record sample for cycle statistics
- if (valid && mCpukHz > 0) {
- double cycles = wcNs * cpukHz * 0.000001;
- mHzStats.sample(cycles);
- }
- unsigned n = mWcStats.n();
- // mCpuUsage.elapsed() is expensive, so don't call it every loop
- if ((n & 127) == 1) {
- long long elapsed = mCpuUsage.elapsed();
- if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
- double perLoop = elapsed / (double) n;
- double perLoop100 = perLoop * 0.01;
- double perLoop1k = perLoop * 0.001;
- double mean = mWcStats.mean();
- double stddev = mWcStats.stddev();
- double minimum = mWcStats.minimum();
- double maximum = mWcStats.maximum();
- double meanCycles = mHzStats.mean();
- double stddevCycles = mHzStats.stddev();
- double minCycles = mHzStats.minimum();
- double maxCycles = mHzStats.maximum();
- mCpuUsage.resetElapsed();
- mWcStats.reset();
- mHzStats.reset();
- ALOGD("CPU usage for %s over past %.1f secs\n"
- " (%u mixer loops at %.1f mean ms per loop):\n"
- " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n"
- " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n"
- " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f",
- title.string(),
- elapsed * .000000001, n, perLoop * .000001,
- mean * .001,
- stddev * .001,
- minimum * .001,
- maximum * .001,
- mean / perLoop100,
- stddev / perLoop100,
- minimum / perLoop100,
- maximum / perLoop100,
- meanCycles / perLoop1k,
- stddevCycles / perLoop1k,
- minCycles / perLoop1k,
- maxCycles / perLoop1k);
- }
- }
- #endif
- };
- // ----------------------------------------------------------------------------
- // ThreadBase
- // ----------------------------------------------------------------------------
- AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
- audio_devices_t outDevice, audio_devices_t inDevice, type_t type)
- : Thread(false /*canCallJava*/),
- mType(type),
- mAudioFlinger(audioFlinger),
- // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize
- // are set by PlaybackThread::readOutputParameters_l() or
- // RecordThread::readInputParameters_l()
- //FIXME: mStandby should be true here. Is this some kind of hack?
- mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
- mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
- // mName will be set by concrete (non-virtual) subclass
- mDeathRecipient(new PMDeathRecipient(this))
- {
- }
- AudioFlinger::ThreadBase::~ThreadBase()
- {
- // mConfigEvents should be empty, but just in case it isn't, free the memory it owns
- mConfigEvents.clear();
- // do not lock the mutex in destructor
- releaseWakeLock_l();
- if (mPowerManager != 0) {
- sp<IBinder> binder = mPowerManager->asBinder();
- binder->unlinkToDeath(mDeathRecipient);
- }
- }
- status_t AudioFlinger::ThreadBase::readyToRun()
- {
- status_t status = initCheck();
- if (status == NO_ERROR) {
- ALOGI("AudioFlinger's thread %p ready to run", this);
- } else {
- ALOGE("No working audio driver found.");
- }
- return status;
- }
- void AudioFlinger::ThreadBase::exit()
- {
- ALOGV("ThreadBase::exit");
- // do any cleanup required for exit to succeed
- preExit();
- {
- // This lock prevents the following race in thread (uniprocessor for illustration):
- // if (!exitPending()) {
- // // context switch from here to exit()
- // // exit() calls requestExit(), what exitPending() observes
- // // exit() calls signal(), which is dropped since no waiters
- // // context switch back from exit() to here
- // mWaitWorkCV.wait(...);
- // // now thread is hung
- // }
- AutoMutex lock(mLock);
- requestExit();
- mWaitWorkCV.broadcast();
- }
- // When Thread::requestExitAndWait is made virtual and this method is renamed to
- // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();"
- requestExitAndWait();
- }
- status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
- {
- status_t status;
- ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
- Mutex::Autolock _l(mLock);
- return sendSetParameterConfigEvent_l(keyValuePairs);
- }
- // sendConfigEvent_l() must be called with ThreadBase::mLock held
- // Can temporarily release the lock if waiting for a reply from processConfigEvents_l().
- status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event)
- {
- status_t status = NO_ERROR;
- mConfigEvents.add(event);
- ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType);
- mWaitWorkCV.signal();
- mLock.unlock();
- {
- Mutex::Autolock _l(event->mLock);
- while (event->mWaitStatus) {
- if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) {
- event->mStatus = TIMED_OUT;
- event->mWaitStatus = false;
- }
- }
- status = event->mStatus;
- }
- mLock.lock();
- return status;
- }
- #ifdef QCOM_DIRECTTRACK
- void AudioFlinger::ThreadBase::effectConfigChanged() {
- ALOGV("New effect is being added to LPA chain, Notifying LPA Direct Track");
- mAudioFlinger->audioConfigChanged(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL);
- }
- #endif
- void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param)
- {
- Mutex::Autolock _l(mLock);
- sendIoConfigEvent_l(event, param);
- }
- // sendIoConfigEvent_l() must be called with ThreadBase::mLock held
- void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param)
- {
- sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param);
- sendConfigEvent_l(configEvent);
- }
- // sendPrioConfigEvent_l() must be called with ThreadBase::mLock held
- void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio)
- {
- sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio);
- sendConfigEvent_l(configEvent);
- }
- // sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held
- status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair)
- {
- sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair);
- return sendConfigEvent_l(configEvent);
- }
- status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent(
- const struct audio_patch *patch,
- audio_patch_handle_t *handle)
- {
- Mutex::Autolock _l(mLock);
- sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle);
- status_t status = sendConfigEvent_l(configEvent);
- if (status == NO_ERROR) {
- CreateAudioPatchConfigEventData *data =
- (CreateAudioPatchConfigEventData *)configEvent->mData.get();
- *handle = data->mHandle;
- }
- return status;
- }
- status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent(
- const audio_patch_handle_t handle)
- {
- Mutex::Autolock _l(mLock);
- sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle);
- return sendConfigEvent_l(configEvent);
- }
- // post condition: mConfigEvents.isEmpty()
- void AudioFlinger::ThreadBase::processConfigEvents_l()
- {
- bool configChanged = false;
- while (!mConfigEvents.isEmpty()) {
- ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size());
- sp<ConfigEvent> event = mConfigEvents[0];
- mConfigEvents.removeAt(0);
- switch (event->mType) {
- case CFG_EVENT_PRIO: {
- PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get();
- // FIXME Need to understand why this has to be done asynchronously
- int err = requestPriority(data->mPid, data->mTid, data->mPrio,
- true /*asynchronous*/);
- if (err != 0) {
- ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
- data->mPrio, data->mPid, data->mTid, err);
- }
- } break;
- case CFG_EVENT_IO: {
- IoConfigEventData *data = (IoConfigEventData *)event->mData.get();
- audioConfigChanged(data->mEvent, data->mParam);
- } break;
- case CFG_EVENT_SET_PARAMETER: {
- SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get();
- if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) {
- configChanged = true;
- }
- } break;
- case CFG_EVENT_CREATE_AUDIO_PATCH: {
- CreateAudioPatchConfigEventData *data =
- (CreateAudioPatchConfigEventData *)event->mData.get();
- event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle);
- } break;
- case CFG_EVENT_RELEASE_AUDIO_PATCH: {
- ReleaseAudioPatchConfigEventData *data =
- (ReleaseAudioPatchConfigEventData *)event->mData.get();
- event->mStatus = releaseAudioPatch_l(data->mHandle);
- } break;
- default:
- ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType);
- break;
- }
- {
- Mutex::Autolock _l(event->mLock);
- if (event->mWaitStatus) {
- event->mWaitStatus = false;
- event->mCond.signal();
- }
- }
- ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this);
- }
- if (configChanged) {
- cacheParameters_l();
- }
- }
- String8 channelMaskToString(audio_channel_mask_t mask, bool output) {
- String8 s;
- if (output) {
- if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, ");
- if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, ");
- if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, ");
- if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, ");
- if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, ");
- if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, ");
- if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, ");
- if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, ");
- if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, ");
- if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, ");
- if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, ");
- if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,");
- if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, ");
- if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, ");
- if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, ");
- if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, ");
- if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " );
- if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " );
- if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, ");
- } else {
- if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, ");
- if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, ");
- if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, ");
- if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, ");
- if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, ");
- if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, ");
- if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, ");
- if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, ");
- if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, ");
- if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, ");
- if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, ");
- if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, ");
- if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, ");
- if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, ");
- if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, ");
- }
- int len = s.length();
- if (s.length() > 2) {
- char *str = s.lockBuffer(len);
- s.unlockBuffer(len - 2);
- }
- return s;
- }
- void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused)
- {
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
- bool locked = AudioFlinger::dumpTryLock(mLock);
- if (!locked) {
- dprintf(fd, "thread %p maybe dead locked\n", this);
- }
- dprintf(fd, " I/O handle: %d\n", mId);
- dprintf(fd, " TID: %d\n", getTid());
- dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no");
- dprintf(fd, " Sample rate: %u\n", mSampleRate);
- dprintf(fd, " HAL frame count: %zu\n", mFrameCount);
- dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize);
- dprintf(fd, " Channel Count: %u\n", mChannelCount);
- dprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask,
- channelMaskToString(mChannelMask, mType != RECORD).string());
- dprintf(fd, " Format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat));
- dprintf(fd, " Frame size: %zu\n", mFrameSize);
- dprintf(fd, " Pending config events:");
- size_t numConfig = mConfigEvents.size();
- if (numConfig) {
- for (size_t i = 0; i < numConfig; i++) {
- mConfigEvents[i]->dump(buffer, SIZE);
- dprintf(fd, "\n %s", buffer);
- }
- dprintf(fd, "\n");
- } else {
- dprintf(fd, " none\n");
- }
- if (locked) {
- mLock.unlock();
- }
- }
- void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
- {
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
- size_t numEffectChains = mEffectChains.size();
- snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains);
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < numEffectChains; ++i) {
- sp<EffectChain> chain = mEffectChains[i];
- if (chain != 0) {
- chain->dump(fd, args);
- }
- }
- }
- void AudioFlinger::ThreadBase::acquireWakeLock(int uid)
- {
- Mutex::Autolock _l(mLock);
- acquireWakeLock_l(uid);
- }
- String16 AudioFlinger::ThreadBase::getWakeLockTag()
- {
- switch (mType) {
- case MIXER:
- return String16("AudioMix");
- case DIRECT:
- return String16("AudioDirectOut");
- case DUPLICATING:
- return String16("AudioDup");
- case RECORD:
- return String16("AudioIn");
- case OFFLOAD:
- return String16("AudioOffload");
- default:
- ALOG_ASSERT(false);
- return String16("AudioUnknown");
- }
- }
- void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid)
- {
- getPowerManager_l();
- if (mPowerManager != 0) {
- sp<IBinder> binder = new BBinder();
- status_t status;
- if (uid >= 0) {
- status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK,
- binder,
- getWakeLockTag(),
- String16("media"),
- uid,
- true /* FIXME force oneway contrary to .aidl */);
- } else {
- status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
- binder,
- getWakeLockTag(),
- String16("media"),
- true /* FIXME force oneway contrary to .aidl */);
- }
- if (status == NO_ERROR) {
- mWakeLockToken = binder;
- }
- ALOGV("acquireWakeLock_l() %s status %d", mName, status);
- }
- }
- void AudioFlinger::ThreadBase::releaseWakeLock()
- {
- Mutex::Autolock _l(mLock);
- releaseWakeLock_l();
- }
- void AudioFlinger::ThreadBase::releaseWakeLock_l()
- {
- if (mWakeLockToken != 0) {
- ALOGV("releaseWakeLock_l() %s", mName);
- if (mPowerManager != 0) {
- mPowerManager->releaseWakeLock(mWakeLockToken, 0,
- true /* FIXME force oneway contrary to .aidl */);
- }
- mWakeLockToken.clear();
- }
- }
- void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) {
- Mutex::Autolock _l(mLock);
- updateWakeLockUids_l(uids);
- }
- void AudioFlinger::ThreadBase::getPowerManager_l() {
- if (mPowerManager == 0) {
- // use checkService() to avoid blocking if power service is not up yet
- sp<IBinder> binder =
- defaultServiceManager()->checkService(String16("power"));
- if (binder == 0) {
- ALOGW("Thread %s cannot connect to the power manager service", mName);
- } else {
- mPowerManager = interface_cast<IPowerManager>(binder);
- binder->linkToDeath(mDeathRecipient);
- }
- }
- }
- void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) {
- getPowerManager_l();
- if (mWakeLockToken == NULL) {
- ALOGE("no wake lock to update!");
- return;
- }
- if (mPowerManager != 0) {
- sp<IBinder> binder = new BBinder();
- status_t status;
- status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(),
- true /* FIXME force oneway contrary to .aidl */);
- ALOGV("acquireWakeLock_l() %s status %d", mName, status);
- }
- }
- void AudioFlinger::ThreadBase::clearPowerManager()
- {
- Mutex::Autolock _l(mLock);
- releaseWakeLock_l();
- mPowerManager.clear();
- }
- void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused)
- {
- sp<ThreadBase> thread = mThread.promote();
- if (thread != 0) {
- thread->clearPowerManager();
- }
- ALOGW("power manager service died !!!");
- }
- void AudioFlinger::ThreadBase::setEffectSuspended(
- const effect_uuid_t *type, bool suspend, int sessionId)
- {
- Mutex::Autolock _l(mLock);
- setEffectSuspended_l(type, suspend, sessionId);
- }
- void AudioFlinger::ThreadBase::setEffectSuspended_l(
- const effect_uuid_t *type, bool suspend, int sessionId)
- {
- sp<EffectChain> chain = getEffectChain_l(sessionId);
- if (chain != 0) {
- if (type != NULL) {
- chain->setEffectSuspended_l(type, suspend);
- } else {
- chain->setEffectSuspendedAll_l(suspend);
- }
- }
- updateSuspendedSessions_l(type, suspend, sessionId);
- }
- void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
- {
- ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId());
- if (index < 0) {
- return;
- }
- const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects =
- mSuspendedSessions.valueAt(index);
- for (size_t i = 0; i < sessionEffects.size(); i++) {
- sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
- for (int j = 0; j < desc->mRefCount; j++) {
- if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
- chain->setEffectSuspendedAll_l(true);
- } else {
- ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
- desc->mType.timeLow);
- chain->setEffectSuspended_l(&desc->mType, true);
- }
- }
- }
- }
- void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
- bool suspend,
- int sessionId)
- {
- ssize_t index = mSuspendedSessions.indexOfKey(sessionId);
- KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
- if (suspend) {
- if (index >= 0) {
- sessionEffects = mSuspendedSessions.valueAt(index);
- } else {
- mSuspendedSessions.add(sessionId, sessionEffects);
- }
- } else {
- if (index < 0) {
- return;
- }
- sessionEffects = mSuspendedSessions.valueAt(index);
- }
- int key = EffectChain::kKeyForSuspendAll;
- if (type != NULL) {
- key = type->timeLow;
- }
- index = sessionEffects.indexOfKey(key);
- sp<SuspendedSessionDesc> desc;
- if (suspend) {
- if (index >= 0) {
- desc = sessionEffects.valueAt(index);
- } else {
- desc = new SuspendedSessionDesc();
- if (type != NULL) {
- desc->mType = *type;
- }
- sessionEffects.add(key, desc);
- ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
- }
- desc->mRefCount++;
- } else {
- if (index < 0) {
- return;
- }
- desc = sessionEffects.valueAt(index);
- if (--desc->mRefCount == 0) {
- ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
- sessionEffects.removeItemsAt(index);
- if (sessionEffects.isEmpty()) {
- ALOGV("updateSuspendedSessions_l() restore removing session %d",
- sessionId);
- mSuspendedSessions.removeItem(sessionId);
- }
- }
- }
- if (!sessionEffects.isEmpty()) {
- mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
- }
- }
- void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
- bool enabled,
- int sessionId)
- {
- Mutex::Autolock _l(mLock);
- checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
- }
- void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
- bool enabled,
- int sessionId)
- {
- if (mType != RECORD) {
- // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
- // another session. This gives the priority to well behaved effect control panels
- // and applications not using global effects.
- // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect
- // global effects
- if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) {
- setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
- }
- }
- sp<EffectChain> chain = getEffectChain_l(sessionId);
- if (chain != 0) {
- chain->checkSuspendOnEffectEnabled(effect, enabled);
- }
- }
- // ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held
- sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
- const sp<AudioFlinger::Client>& client,
- const sp<IEffectClient>& effectClient,
- int32_t priority,
- int sessionId,
- effect_descriptor_t *desc,
- int *enabled,
- status_t *status)
- {
- sp<EffectModule> effect;
- sp<EffectHandle> handle;
- status_t lStatus;
- sp<EffectChain> chain;
- bool chainCreated = false;
- bool effectCreated = false;
- bool effectRegistered = false;
- lStatus = initCheck();
- if (lStatus != NO_ERROR) {
- ALOGW("createEffect_l() Audio driver not initialized.");
- goto Exit;
- }
- // Reject any effect on Direct output threads for now, since the format of
- // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo).
- if (mType == DIRECT) {
- ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s",
- desc->name, mName);
- lStatus = BAD_VALUE;
- goto Exit;
- }
- // Reject any effect on mixer or duplicating multichannel sinks.
- // TODO: fix both format and multichannel issues with effects.
- if ((mType == MIXER || mType == DUPLICATING) && mChannelCount > FCC_2) {
- ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads",
- desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING");
- lStatus = BAD_VALUE;
- goto Exit;
- }
- // Allow global effects only on offloaded and mixer threads
- if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
- switch (mType) {
- case MIXER:
- case OFFLOAD:
- break;
- case DIRECT:
- case DUPLICATING:
- case RECORD:
- default:
- ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName);
- lStatus = BAD_VALUE;
- goto Exit;
- }
- }
- // Only Pre processor effects are allowed on input threads and only on input threads
- if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
- ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
- desc->name, desc->flags, mType);
- lStatus = BAD_VALUE;
- goto Exit;
- }
- ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
- { // scope for mLock
- Mutex::Autolock _l(mLock);
- // check for existing effect chain with the requested audio session
- chain = getEffectChain_l(sessionId);
- if (chain == 0) {
- // create a new chain for this session
- ALOGV("createEffect_l() new effect chain for session %d", sessionId);
- chain = new EffectChain(this, sessionId);
- addEffectChain_l(chain);
- chain->setStrategy(getStrategyForSession_l(sessionId));
- chainCreated = true;
- #ifdef QCOM_DIRECTTRACK
- if(sessionId == mAudioFlinger->mLPASessionId) {
- // Clear reference to previous effect chain if any
- if(mAudioFlinger->mLPAEffectChain.get()) {
- mAudioFlinger->mLPAEffectChain.clear();
- }
- ALOGV("New EffectChain is created for LPA session ID %d", sessionId);
- mAudioFlinger->mLPAEffectChain = chain;
- chain->setLPAFlag(true);
- // For LPA, the volume will be applied in DSP. No need for volume
- // control in the Effect chain, so setting it to unity.
- uint32_t volume = 0x1000000; // Equals to 1.0 in 8.24 format
- chain->setVolume_l(&volume,&volume);
- }
- #endif
- } else {
- effect = chain->getEffectFromDesc_l(desc);
- }
- ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get());
- if (effect == 0) {
- int id = mAudioFlinger->nextUniqueId();
- // Check CPU and memory usage
- lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
- if (lStatus != NO_ERROR) {
- goto Exit;
- }
- effectRegistered = true;
- // create a new effect module if none present in the chain
- effect = new EffectModule(this, chain, desc, id, sessionId);
- lStatus = effect->status();
- if (lStatus != NO_ERROR) {
- goto Exit;
- }
- effect->setOffloaded(mType == OFFLOAD, mId);
- lStatus = chain->addEffect_l(effect);
- if (lStatus != NO_ERROR) {
- goto Exit;
- }
- effectCreated = true;
- #ifdef QCOM_DIRECTTRACK
- effect->setDevice(mAudioFlinger->mLPASessionId == sessionId ? mAudioFlinger->mDirectDevice:mOutDevice);
- #else
- effect->setDevice(mOutDevice);
- #endif
- effect->setDevice(mInDevice);
- effect->setMode(mAudioFlinger->getMode());
- effect->setAudioSource(mAudioSource);
- }
- // create effect handle and connect it to effect module
- handle = new EffectHandle(effect, client, effectClient, priority);
- lStatus = handle->initCheck();
- if (lStatus == OK) {
- lStatus = effect->addHandle(handle.get());
- }
- if (enabled != NULL) {
- *enabled = (int)effect->isEnabled();
- }
- }
- Exit:
- if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
- Mutex::Autolock _l(mLock);
- if (effectCreated) {
- chain->removeEffect_l(effect);
- }
- if (effectRegistered) {
- AudioSystem::unregisterEffect(effect->id());
- }
- if (chainCreated) {
- removeEffectChain_l(chain);
- }
- handle.clear();
- }
- *status = lStatus;
- return handle;
- }
- sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId)
- {
- Mutex::Autolock _l(mLock);
- return getEffect_l(sessionId, effectId);
- }
- sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
- {
- sp<EffectChain> chain = getEffectChain_l(sessionId);
- return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
- }
- // PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
- // PlaybackThread::mLock held
- status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
- {
- // check for existing effect chain with the requested audio session
- int sessionId = effect->sessionId();
- sp<EffectChain> chain = getEffectChain_l(sessionId);
- bool chainCreated = false;
- ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(),
- "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x",
- this, effect->desc().name, effect->desc().flags);
- if (chain == 0) {
- // create a new chain for this session
- ALOGV("addEffect_l() new effect chain for session %d", sessionId);
- chain = new EffectChain(this, sessionId);
- addEffectChain_l(chain);
- chain->setStrategy(getStrategyForSession_l(sessionId));
- chainCreated = true;
- }
- ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
- if (chain->getEffectFromId_l(effect->id()) != 0) {
- ALOGW("addEffect_l() %p effect %s already present in chain %p",
- this, effect->desc().name, chain.get());
- return BAD_VALUE;
- }
- effect->setOffloaded(mType == OFFLOAD, mId);
- status_t status = chain->addEffect_l(effect);
- if (status != NO_ERROR) {
- if (chainCreated) {
- removeEffectChain_l(chain);
- }
- return status;
- }
- effect->setDevice(mOutDevice);
- effect->setDevice(mInDevice);
- effect->setMode(mAudioFlinger->getMode());
- effect->setAudioSource(mAudioSource);
- return NO_ERROR;
- }
- void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
- ALOGV("removeEffect_l() %p effect %p", this, effect.get());
- effect_descriptor_t desc = effect->desc();
- if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
- detachAuxEffect_l(effect->id());
- }
- sp<EffectChain> chain = effect->chain().promote();
- if (chain != 0) {
- // remove effect chain if removing last effect
- if (chain->removeEffect_l(effect) == 0) {
- removeEffectChain_l(chain);
- }
- } else {
- ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
- }
- }
- void AudioFlinger::ThreadBase::lockEffectChains_l(
- Vector< sp<AudioFlinger::EffectChain> >& effectChains)
- {
- effectChains = mEffectChains;
- for (size_t i = 0; i < mEffectChains.size(); i++) {
- mEffectChains[i]->lock();
- }
- }
- void AudioFlinger::ThreadBase::unlockEffectChains(
- const Vector< sp<AudioFlinger::EffectChain> >& effectChains)
- {
- for (size_t i = 0; i < effectChains.size(); i++) {
- effectChains[i]->unlock();
- }
- }
- sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
- {
- Mutex::Autolock _l(mLock);
- return getEffectChain_l(sessionId);
- }
- sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const
- {
- size_t size = mEffectChains.size();
- for (size_t i = 0; i < size; i++) {
- if (mEffectChains[i]->sessionId() == sessionId) {
- return mEffectChains[i];
- }
- }
- return 0;
- }
- void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
- {
- Mutex::Autolock _l(mLock);
- size_t size = mEffectChains.size();
- for (size_t i = 0; i < size; i++) {
- mEffectChains[i]->setMode_l(mode);
- }
- }
- void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config)
- {
- config->type = AUDIO_PORT_TYPE_MIX;
- config->ext.mix.handle = mId;
- config->sample_rate = mSampleRate;
- config->format = mFormat;
- config->channel_mask = mChannelMask;
- config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
- AUDIO_PORT_CONFIG_FORMAT;
- }
- // ----------------------------------------------------------------------------
- // Playback
- // ----------------------------------------------------------------------------
- AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
- AudioStreamOut* output,
- audio_io_handle_t id,
- audio_devices_t device,
- type_t type)
- : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
- mNormalFrameCount(0), mSinkBuffer(NULL),
- mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
- mMixerBuffer(NULL),
- mMixerBufferSize(0),
- mMixerBufferFormat(AUDIO_FORMAT_INVALID),
- mMixerBufferValid(false),
- mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
- mEffectBuffer(NULL),
- mEffectBufferSize(0),
- mEffectBufferFormat(AUDIO_FORMAT_INVALID),
- mEffectBufferValid(false),
- mSuspended(0), mBytesWritten(0),
- mActiveTracksGeneration(0),
- // mStreamTypes[] initialized in constructor body
- mOutput(output),
- mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
- mMixerStatus(MIXER_IDLE),
- mMixerStatusIgnoringFastTracks(MIXER_IDLE),
- standbyDelay(AudioFlinger::mStandbyTimeInNsecs),
- mBytesRemaining(0),
- mCurrentWriteLength(0),
- mUseAsyncWrite(false),
- mWriteAckSequence(0),
- mDrainSequence(0),
- mSignalPending(false),
- mScreenState(AudioFlinger::mScreenState),
- // index 0 is reserved for normal mixer's submix
- mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1),
- // mLatchD, mLatchQ,
- mLatchDValid(false), mLatchQValid(false)
- {
- snprintf(mName, kNameLength, "AudioOut_%X", id);
- mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName);
- // Assumes constructor is called by AudioFlinger with it's mLock held, but
- // it would be safer to explicitly pass initial masterVolume/masterMute as
- // parameter.
- //
- // If the HAL we are using has support for master volume or master mute,
- // then do not attenuate or mute during mixing (just leave the volume at 1.0
- // and the mute set to false).
- mMasterVolume = audioFlinger->masterVolume_l();
- mMasterMute = audioFlinger->masterMute_l();
- if (mOutput && mOutput->audioHwDev) {
- if (mOutput->audioHwDev->canSetMasterVolume()) {
- mMasterVolume = 1.0;
- }
- if (mOutput->audioHwDev->canSetMasterMute()) {
- mMasterMute = false;
- }
- }
- readOutputParameters_l();
- // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
- // There is no AUDIO_STREAM_MIN, and ++ operator does not compile
- for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT;
- stream = (audio_stream_type_t) (stream + 1)) {
- mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream);
- mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream);
- }
- // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here,
- // because mAudioFlinger doesn't have one to copy from
- }
- AudioFlinger::PlaybackThread::~PlaybackThread()
- {
- mAudioFlinger->unregisterWriter(mNBLogWriter);
- free(mSinkBuffer);
- free(mMixerBuffer);
- free(mEffectBuffer);
- }
- void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
- {
- dumpInternals(fd, args);
- dumpTracks(fd, args);
- dumpEffectChains(fd, args);
- }
- void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused)
- {
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
- result.appendFormat(" Stream volumes in dB: ");
- for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
- const stream_type_t *st = &mStreamTypes[i];
- if (i > 0) {
- result.appendFormat(", ");
- }
- result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume));
- if (st->mute) {
- result.append("M");
- }
- }
- result.append("\n");
- write(fd, result.string(), result.length());
- result.clear();
- // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way.
- FastTrackUnderruns underruns = getFastTrackUnderruns(0);
- dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n",
- underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty);
- size_t numtracks = mTracks.size();
- size_t numactive = mActiveTracks.size();
- dprintf(fd, " %d Tracks", numtracks);
- size_t numactiveseen = 0;
- if (numtracks) {
- dprintf(fd, " of which %d are active\n", numactive);
- Track::appendDumpHeader(result);
- for (size_t i = 0; i < numtracks; ++i) {
- sp<Track> track = mTracks[i];
- if (track != 0) {
- bool active = mActiveTracks.indexOf(track) >= 0;
- if (active) {
- numactiveseen++;
- }
- track->dump(buffer, SIZE, active);
- result.append(buffer);
- }
- }
- } else {
- result.append("\n");
- }
- if (numactiveseen != numactive) {
- // some tracks in the active list were not in the tracks list
- snprintf(buffer, SIZE, " The following tracks are in the active list but"
- " not in the track list\n");
- result.append(buffer);
- Track::appendDumpHeader(result);
- for (size_t i = 0; i < numactive; ++i) {
- sp<Track> track = mActiveTracks[i].promote();
- if (track != 0 && mTracks.indexOf(track) < 0) {
- track->dump(buffer, SIZE, true);
- result.append(buffer);
- }
- }
- }
- write(fd, result.string(), result.size());
- }
- void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
- {
- dprintf(fd, "\nOutput thread %p:\n", this);
- dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount);
- dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
- dprintf(fd, " Total writes: %d\n", mNumWrites);
- dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites);
- dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no");
- dprintf(fd, " Suspend count: %d\n", mSuspended);
- dprintf(fd, " Sink buffer : %p\n", mSinkBuffer);
- dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer);
- dprintf(fd, " Effect buffer: %p\n", mEffectBuffer);
- dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask);
- dumpBase(fd, args);
- }
- // Thread virtuals
- void AudioFlinger::PlaybackThread::onFirstRef()
- {
- run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
- }
- // ThreadBase virtuals
- void AudioFlinger::PlaybackThread::preExit()
- {
- ALOGV(" preExit()");
- // FIXME this is using hard-coded strings but in the future, this functionality will be
- // converted to use audio HAL extensions required to support tunneling
- mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1");
- }
- // PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
- sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
- const sp<AudioFlinger::Client>& client,
- audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- size_t *pFrameCount,
- const sp<IMemory>& sharedBuffer,
- int sessionId,
- IAudioFlinger::track_flags_t *flags,
- pid_t tid,
- int uid,
- status_t *status)
- {
- size_t frameCount = *pFrameCount;
- sp<Track> track;
- status_t lStatus;
- bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0;
- // client expresses a preference for FAST, but we get the final say
- if (*flags & IAudioFlinger::TRACK_FAST) {
- if (
- // not timed
- (!isTimed) &&
- // either of these use cases:
- (
- // use case 1: shared buffer with any frame count
- (
- (sharedBuffer != 0)
- ) ||
- // use case 2: callback handler and frame count is default or at least as large as HAL
- (
- (tid != -1) &&
- ((frameCount == 0) ||
- (frameCount >= mFrameCount))
- )
- ) &&
- // PCM data
- audio_is_linear_pcm(format) &&
- // identical channel mask to sink, or mono in and stereo sink
- (channelMask == mChannelMask ||
- (channelMask == AUDIO_CHANNEL_OUT_MONO &&
- mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) &&
- // hardware sample rate
- (sampleRate == mSampleRate) &&
- // normal mixer has an associated fast mixer
- hasFastMixer() &&
- // there are sufficient fast track slots available
- (mFastTrackAvailMask != 0)
- // FIXME test that MixerThread for this fast track has a capable output HAL
- // FIXME add a permission test also?
- ) {
- // if frameCount not specified, then it defaults to fast mixer (HAL) frame count
- if (frameCount == 0) {
- // read the fast track multiplier property the first time it is needed
- int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit);
- if (ok != 0) {
- ALOGE("%s pthread_once failed: %d", __func__, ok);
- }
- frameCount = mFrameCount * sFastTrackMultiplier;
- }
- ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d",
- frameCount, mFrameCount);
- } else {
- ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d "
- "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x "
- "sampleRate=%u mSampleRate=%u "
- "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
- isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat,
- audio_is_linear_pcm(format),
- channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
- *flags &= ~IAudioFlinger::TRACK_FAST;
- // For compatibility with AudioTrack calculation, buffer depth is forced
- // to be at least 2 x the normal mixer frame count and cover audio hardware latency.
- // This is probably too conservative, but legacy application code may depend on it.
- // If you change this calculation, also review the start threshold which is related.
- uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream);
- uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
- if (minBufCount < 2) {
- minBufCount = 2;
- }
- size_t minFrameCount = mNormalFrameCount * minBufCount;
- if (frameCount < minFrameCount) {
- frameCount = minFrameCount;
- }
- }
- }
- *pFrameCount = frameCount;
- switch (mType) {
- case DIRECT:
- if (audio_is_linear_pcm(format) ||
- audio_is_compress_voip_format(format) ||
- audio_is_compress_capture_format(format)) {
- if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
- ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x "
- "for output %p with format %#x",
- sampleRate, format, channelMask, mOutput, mFormat);
- lStatus = BAD_VALUE;
- goto Exit;
- }
- }
- break;
- case OFFLOAD:
- if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
- ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \""
- "for output %p with format %#x",
- sampleRate, format, channelMask, mOutput, mFormat);
- lStatus = BAD_VALUE;
- goto Exit;
- }
- break;
- default:
- if (!audio_is_linear_pcm(format)) {
- ALOGE("createTrack_l() Bad parameter: format %#x \""
- "for output %p with format %#x",
- format, mOutput, mFormat);
- lStatus = BAD_VALUE;
- goto Exit;
- }
- if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
- ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate);
- lStatus = BAD_VALUE;
- goto Exit;
- }
- break;
- }
- lStatus = initCheck();
- if (lStatus != NO_ERROR) {
- ALOGE("createTrack_l() audio driver not initialized");
- goto Exit;
- }
- { // scope for mLock
- Mutex::Autolock _l(mLock);
- // all tracks in same audio session must share the same routing strategy otherwise
- // conflicts will happen when tracks are moved from one output to another by audio policy
- // manager
- uint32_t strategy = AudioSystem::getStrategyForStream(streamType);
- for (size_t i = 0; i < mTracks.size(); ++i) {
- sp<Track> t = mTracks[i];
- if (t != 0 && t->isExternalTrack()) {
- uint32_t actual = AudioSystem::getStrategyForStream(t->streamType());
- if (sessionId == t->sessionId() && strategy != actual) {
- ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
- strategy, actual);
- lStatus = BAD_VALUE;
- goto Exit;
- }
- }
- }
- if (!isTimed) {
- track = new Track(this, client, streamType, sampleRate, format,
- channelMask, frameCount, NULL, sharedBuffer,
- sessionId, uid, *flags, TrackBase::TYPE_DEFAULT);
- } else {
- track = TimedTrack::create(this, client, streamType, sampleRate, format,
- channelMask, frameCount, sharedBuffer, sessionId, uid);
- }
- // new Track always returns non-NULL,
- // but TimedTrack::create() is a factory that could fail by returning NULL
- lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
- if (lStatus != NO_ERROR) {
- ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus);
- // track must be cleared from the caller as the caller has the AF lock
- goto Exit;
- }
- mTracks.add(track);
- sp<EffectChain> chain = getEffectChain_l(sessionId);
- if (chain != 0) {
- ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
- track->setMainBuffer(chain->inBuffer());
- chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
- chain->incTrackCnt();
- }
- if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) {
- pid_t callingPid = IPCThreadState::self()->getCallingPid();
- // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
- // so ask activity manager to do this on our behalf
- sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp);
- }
- }
- lStatus = NO_ERROR;
- Exit:
- *status = lStatus;
- return track;
- }
- uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const
- {
- return latency;
- }
- uint32_t AudioFlinger::PlaybackThread::latency() const
- {
- Mutex::Autolock _l(mLock);
- return latency_l();
- }
- uint32_t AudioFlinger::PlaybackThread::latency_l() const
- {
- if (initCheck() == NO_ERROR) {
- return correctLatency_l(mOutput->stream->get_latency(mOutput->stream));
- } else {
- return 0;
- }
- }
- void AudioFlinger::PlaybackThread::setMasterVolume(float value)
- {
- Mutex::Autolock _l(mLock);
- // Don't apply master volume in SW if our HAL can do it for us.
- if (mOutput && mOutput->audioHwDev &&
- mOutput->audioHwDev->canSetMasterVolume()) {
- mMasterVolume = 1.0;
- } else {
- mMasterVolume = value;
- }
- }
- void AudioFlinger::PlaybackThread::setMasterMute(bool muted)
- {
- Mutex::Autolock _l(mLock);
- // Don't apply master mute in SW if our HAL can do it for us.
- if (mOutput && mOutput->audioHwDev &&
- mOutput->audioHwDev->canSetMasterMute()) {
- mMasterMute = false;
- } else {
- mMasterMute = muted;
- }
- }
- void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
- {
- Mutex::Autolock _l(mLock);
- mStreamTypes[stream].volume = value;
- broadcast_l();
- }
- void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
- {
- Mutex::Autolock _l(mLock);
- mStreamTypes[stream].mute = muted;
- broadcast_l();
- }
- float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
- {
- Mutex::Autolock _l(mLock);
- return mStreamTypes[stream].volume;
- }
- // addTrack_l() must be called with ThreadBase::mLock held
- status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
- {
- status_t status = ALREADY_EXISTS;
- // set retry count for buffer fill
- track->mRetryCount = kMaxTrackStartupRetries;
- if (mActiveTracks.indexOf(track) < 0) {
- // the track is newly added, make sure it fills up all its
- // buffers before playing. This is to ensure the client will
- // effectively get the latency it requested.
- if (track->isExternalTrack()) {
- TrackBase::track_state state = track->mState;
- mLock.unlock();
- status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId());
- mLock.lock();
- // abort track was stopped/paused while we released the lock
- if (state != track->mState) {
- if (status == NO_ERROR) {
- mLock.unlock();
- AudioSystem::stopOutput(mId, track->streamType(), track->sessionId());
- mLock.lock();
- }
- return INVALID_OPERATION;
- }
- // abort if start is rejected by audio policy manager
- if (status != NO_ERROR) {
- return PERMISSION_DENIED;
- }
- #ifdef ADD_BATTERY_DATA
- // to track the speaker usage
- addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
- #endif
- }
- track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING;
- track->mResetDone = false;
- track->mPresentationCompleteFrames = 0;
- mActiveTracks.add(track);
- mWakeLockUids.add(track->uid());
- mActiveTracksGeneration++;
- mLatestActiveTrack = track;
- sp<EffectChain> chain = getEffectChain_l(track->sessionId());
- if (chain != 0) {
- ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(),
- track->sessionId());
- chain->incActiveTrackCnt();
- }
- status = NO_ERROR;
- }
- onAddNewTrack_l();
- return status;
- }
- bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
- {
- track->terminate();
- // active tracks are removed by threadLoop()
- bool trackActive = (mActiveTracks.indexOf(track) >= 0);
- track->mState = TrackBase::STOPPED;
- if (!trackActive) {
- removeTrack_l(track);
- } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) {
- track->mState = TrackBase::STOPPING_1;
- }
- return trackActive;
- }
- void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
- {
- track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
- mTracks.remove(track);
- deleteTrackName_l(track->name());
- // redundant as track is about to be destroyed, for dumpsys only
- track->mName = -1;
- if (track->isFastTrack()) {
- int index = track->mFastIndex;
- ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks);
- ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index)));
- mFastTrackAvailMask |= 1 << index;
- // redundant as track is about to be destroyed, for dumpsys only
- track->mFastIndex = -1;
- }
- sp<EffectChain> chain = getEffectChain_l(track->sessionId());
- if (chain != 0) {
- chain->decTrackCnt();
- }
- }
- void AudioFlinger::PlaybackThread::broadcast_l()
- {
- // Thread could be blocked waiting for async
- // so signal it to handle state changes immediately
- // If threadLoop is currently unlocked a signal of mWaitWorkCV will
- // be lost so we also flag to prevent it blocking on mWaitWorkCV
- mSignalPending = true;
- mWaitWorkCV.broadcast();
- }
- String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
- {
- Mutex::Autolock _l(mLock);
- if (initCheck() != NO_ERROR) {
- return String8();
- }
- char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
- const String8 out_s8(s);
- free(s);
- return out_s8;
- }
- void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) {
- AudioSystem::OutputDescriptor desc;
- void *param2 = NULL;
- ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event,
- param);
- switch (event) {
- case AudioSystem::OUTPUT_OPENED:
- case AudioSystem::OUTPUT_CONFIG_CHANGED:
- desc.channelMask = mChannelMask;
- desc.samplingRate = mSampleRate;
- desc.format = mFormat;
- desc.frameCount = mNormalFrameCount; // FIXME see
- // AudioFlinger::frameCount(audio_io_handle_t)
- desc.latency = latency_l();
- param2 = &desc;
- break;
- case AudioSystem::STREAM_CONFIG_CHANGED:
- param2 = ¶m;
- case AudioSystem::OUTPUT_CLOSED:
- default:
- break;
- }
- mAudioFlinger->audioConfigChanged(event, mId, param2);
- }
- void AudioFlinger::PlaybackThread::writeCallback()
- {
- ALOG_ASSERT(mCallbackThread != 0);
- mCallbackThread->resetWriteBlocked();
- }
- void AudioFlinger::PlaybackThread::drainCallback()
- {
- ALOG_ASSERT(mCallbackThread != 0);
- mCallbackThread->resetDraining();
- }
- void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence)
- {
- Mutex::Autolock _l(mLock);
- // reject out of sequence requests
- if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) {
- mWriteAckSequence &= ~1;
- mWaitWorkCV.signal();
- }
- }
- void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence)
- {
- Mutex::Autolock _l(mLock);
- // reject out of sequence requests
- if ((mDrainSequence & 1) && (sequence == mDrainSequence)) {
- mDrainSequence &= ~1;
- mWaitWorkCV.signal();
- }
- }
- // static
- int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event,
- void *param __unused,
- void *cookie)
- {
- AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie;
- ALOGV("asyncCallback() event %d", event);
- switch (event) {
- case STREAM_CBK_EVENT_WRITE_READY:
- me->writeCallback();
- break;
- case STREAM_CBK_EVENT_DRAIN_READY:
- me->drainCallback();
- break;
- default:
- ALOGW("asyncCallback() unknown event %d", event);
- break;
- }
- return 0;
- }
- void AudioFlinger::PlaybackThread::readOutputParameters_l()
- {
- // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL
- mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
- mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
- if (!audio_is_output_channel(mChannelMask)) {
- LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask);
- }
- if ((mType == MIXER || mType == DUPLICATING)
- && !isValidPcmSinkChannelMask(mChannelMask)) {
- LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output",
- mChannelMask);
- }
- mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
- mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
- mFormat = mHALFormat;
- if (!audio_is_valid_format(mFormat)) {
- LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat);
- }
- if ((mType == MIXER || mType == DUPLICATING)
- && !isValidPcmSinkFormat(mFormat)) {
- LOG_FATAL("HAL format %#x not supported for mixed output",
- mFormat);
- }
- mFrameSize = audio_stream_out_frame_size(mOutput->stream);
- mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common);
- mFrameCount = mBufferSize / mFrameSize;
- if (mFrameCount & 15) {
- ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames",
- mFrameCount);
- }
- if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) &&
- (mOutput->stream->set_callback != NULL)) {
- if (mOutput->stream->set_callback(mOutput->stream,
- AudioFlinger::PlaybackThread::asyncCallback, this) == 0) {
- mUseAsyncWrite = true;
- mCallbackThread = new AudioFlinger::AsyncCallbackThread(this);
- }
- }
- // Calculate size of normal sink buffer relative to the HAL output buffer size
- double multiplier = 1.0;
- if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
- kUseFastMixer == FastMixer_Dynamic)) {
- size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000;
- size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000;
- // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer
- minNormalFrameCount = (minNormalFrameCount + 15) & ~15;
- maxNormalFrameCount = maxNormalFrameCount & ~15;
- if (maxNormalFrameCount < minNormalFrameCount) {
- maxNormalFrameCount = minNormalFrameCount;
- }
- multiplier = (double) minNormalFrameCount / (double) mFrameCount;
- if (multiplier <= 1.0) {
- multiplier = 1.0;
- } else if (multiplier <= 2.0) {
- if (2 * mFrameCount <= maxNormalFrameCount) {
- multiplier = 2.0;
- } else {
- multiplier = (double) maxNormalFrameCount / (double) mFrameCount;
- }
- } else {
- // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL
- // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast
- // track, but we sometimes have to do this to satisfy the maximum frame count
- // constraint)
- // FIXME this rounding up should not be done if no HAL SRC
- uint32_t truncMult = (uint32_t) multiplier;
- if ((truncMult & 1)) {
- if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) {
- ++truncMult;
- }
- }
- multiplier = (double) truncMult;
- }
- }
- mNormalFrameCount = multiplier * mFrameCount;
- // round up to nearest 16 frames to satisfy AudioMixer
- if (mType == MIXER || mType == DUPLICATING) {
- mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
- }
- ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount,
- mNormalFrameCount);
- // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames.
- // Originally this was int16_t[] array, need to remove legacy implications.
- free(mSinkBuffer);
- mSinkBuffer = NULL;
- // For sink buffer size, we use the frame size from the downstream sink to avoid problems
- // with non PCM formats for compressed music, e.g. AAC, and Offload threads.
- const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
- (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
- // We resize the mMixerBuffer according to the requirements of the sink buffer which
- // drives the output.
- free(mMixerBuffer);
- mMixerBuffer = NULL;
- if (mMixerBufferEnabled) {
- mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT.
- mMixerBufferSize = mNormalFrameCount * mChannelCount
- * audio_bytes_per_sample(mMixerBufferFormat);
- (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize);
- }
- free(mEffectBuffer);
- mEffectBuffer = NULL;
- if (mEffectBufferEnabled) {
- mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only
- mEffectBufferSize = mNormalFrameCount * mChannelCount
- * audio_bytes_per_sample(mEffectBufferFormat);
- (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize);
- }
- // force reconfiguration of effect chains and engines to take new buffer size and audio
- // parameters into account
- // Note that mLock is not held when readOutputParameters_l() is called from the constructor
- // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
- // matter.
- // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
- Vector< sp<EffectChain> > effectChains = mEffectChains;
- for (size_t i = 0; i < effectChains.size(); i ++) {
- mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
- }
- }
- status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
- {
- if (halFrames == NULL || dspFrames == NULL) {
- return BAD_VALUE;
- }
- Mutex::Autolock _l(mLock);
- if (initCheck() != NO_ERROR) {
- return INVALID_OPERATION;
- }
- size_t framesWritten = mBytesWritten / mFrameSize;
- *halFrames = framesWritten;
- if (isSuspended()) {
- // return an estimation of rendered frames when the output is suspended
- size_t latencyFrames = (latency_l() * mSampleRate) / 1000;
- *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0;
- return NO_ERROR;
- } else {
- status_t status;
- uint32_t frames;
- status = mOutput->stream->get_render_position(mOutput->stream, &frames);
- *dspFrames = (size_t)frames;
- return status;
- }
- }
- uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const
- {
- Mutex::Autolock _l(mLock);
- uint32_t result = 0;
- if (getEffectChain_l(sessionId) != 0) {
- result = EFFECT_SESSION;
- }
- for (size_t i = 0; i < mTracks.size(); ++i) {
- sp<Track> track = mTracks[i];
- if (sessionId == track->sessionId() && !track->isInvalid()) {
- result |= TRACK_SESSION;
- break;
- }
- }
- return result;
- }
- uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
- {
- // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
- // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
- if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
- return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
- }
- for (size_t i = 0; i < mTracks.size(); i++) {
- sp<Track> track = mTracks[i];
- if (sessionId == track->sessionId() && !track->isInvalid()) {
- return AudioSystem::getStrategyForStream(track->streamType());
- }
- }
- return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
- }
- AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
- {
- Mutex::Autolock _l(mLock);
- return mOutput;
- }
- AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
- {
- Mutex::Autolock _l(mLock);
- AudioStreamOut *output = mOutput;
- mOutput = NULL;
- // FIXME FastMixer might also have a raw ptr to mOutputSink;
- // must push a NULL and wait for ack
- mOutputSink.clear();
- mPipeSink.clear();
- mNormalSink.clear();
- return output;
- }
- // this method must always be called either with ThreadBase mLock held or inside the thread loop
- audio_stream_t* AudioFlinger::PlaybackThread::stream() const
- {
- if (mOutput == NULL) {
- return NULL;
- }
- return &mOutput->stream->common;
- }
- uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const
- {
- return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000);
- }
- status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event)
- {
- if (!isValidSyncEvent(event)) {
- return BAD_VALUE;
- }
- Mutex::Autolock _l(mLock);
- for (size_t i = 0; i < mTracks.size(); ++i) {
- sp<Track> track = mTracks[i];
- if (event->triggerSession() == track->sessionId()) {
- (void) track->setSyncEvent(event);
- return NO_ERROR;
- }
- }
- return NAME_NOT_FOUND;
- }
- bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const
- {
- return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE;
- }
- void AudioFlinger::PlaybackThread::threadLoop_removeTracks(
- const Vector< sp<Track> >& tracksToRemove)
- {
- size_t count = tracksToRemove.size();
- if (count > 0) {
- for (size_t i = 0 ; i < count ; i++) {
- const sp<Track>& track = tracksToRemove.itemAt(i);
- if (track->isExternalTrack()) {
- AudioSystem::stopOutput(mId, track->streamType(), track->sessionId());
- #ifdef ADD_BATTERY_DATA
- // to track the speaker usage
- addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
- #endif
- if (track->isTerminated()) {
- AudioSystem::releaseOutput(mId);
- }
- }
- }
- }
- }
- void AudioFlinger::PlaybackThread::checkSilentMode_l()
- {
- if (!mMasterMute) {
- char value[PROPERTY_VALUE_MAX];
- if (property_get("ro.audio.silent", value, "0") > 0) {
- char *endptr;
- unsigned long ul = strtoul(value, &endptr, 0);
- if (*endptr == '\0' && ul != 0) {
- ALOGD("Silence is golden");
- // The setprop command will not allow a property to be changed after
- // the first time it is set, so we don't have to worry about un-muting.
- setMasterMute_l(true);
- }
- }
- }
- }
- // shared by MIXER and DIRECT, overridden by DUPLICATING
- ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
- {
- // FIXME rewrite to reduce number of system calls
- mLastWriteTime = systemTime();
- mInWrite = true;
- ssize_t bytesWritten;
- const size_t offset = mCurrentWriteLength - mBytesRemaining;
- // If an NBAIO sink is present, use it to write the normal mixer's submix
- if (mNormalSink != 0) {
- const size_t count = mBytesRemaining / mFrameSize;
- ATRACE_BEGIN("write");
- // update the setpoint when AudioFlinger::mScreenState changes
- uint32_t screenState = AudioFlinger::mScreenState;
- if (screenState != mScreenState) {
- mScreenState = screenState;
- MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
- if (pipe != NULL) {
- pipe->setAvgFrames((mScreenState & 1) ?
- (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
- }
- }
- ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count);
- ATRACE_END();
- if (framesWritten > 0) {
- bytesWritten = framesWritten * mFrameSize;
- } else {
- bytesWritten = framesWritten;
- }
- status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp);
- if (status == NO_ERROR) {
- size_t totalFramesWritten = mNormalSink->framesWritten();
- if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) {
- mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition;
- // mLatchD.mFramesReleased is set immediately before D is clocked into Q
- mLatchDValid = true;
- }
- }
- // otherwise use the HAL / AudioStreamOut directly
- } else {
- // Direct output and offload threads
- if (mUseAsyncWrite) {
- ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request");
- mWriteAckSequence += 2;
- mWriteAckSequence |= 1;
- ALOG_ASSERT(mCallbackThread != 0);
- mCallbackThread->setWriteBlocked(mWriteAckSequence);
- }
- // FIXME We should have an implementation of timestamps for direct output threads.
- // They are used e.g for multichannel PCM playback over HDMI.
- bytesWritten = mOutput->stream->write(mOutput->stream,
- (char *)mSinkBuffer + offset, mBytesRemaining);
- if (mUseAsyncWrite &&
- ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
- // do not wait for async callback in case of error of full write
- mWriteAckSequence &= ~1;
- ALOG_ASSERT(mCallbackThread != 0);
- mCallbackThread->setWriteBlocked(mWriteAckSequence);
- }
- }
- mNumWrites++;
- mInWrite = false;
- mStandby = false;
- return bytesWritten;
- }
- void AudioFlinger::PlaybackThread::threadLoop_drain()
- {
- if (mOutput->stream->drain) {
- ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full");
- if (mUseAsyncWrite) {
- ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request");
- mDrainSequence |= 1;
- ALOG_ASSERT(mCallbackThread != 0);
- mCallbackThread->setDraining(mDrainSequence);
- }
- mOutput->stream->drain(mOutput->stream,
- (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY
- : AUDIO_DRAIN_ALL);
- }
- }
- void AudioFlinger::PlaybackThread::threadLoop_exit()
- {
- // Default implementation has nothing to do
- }
- /*
- The derived values that are cached:
- - mSinkBufferSize from frame count * frame size
- - activeSleepTime from activeSleepTimeUs()
- - idleSleepTime from idleSleepTimeUs()
- - standbyDelay from mActiveSleepTimeUs (DIRECT only)
- - maxPeriod from frame count and sample rate (MIXER only)
- The parameters that affect these derived values are:
- - frame count
- - frame size
- - sample rate
- - device type: A2DP or not
- - device latency
- - format: PCM or not
- - active sleep time
- - idle sleep time
- */
- void AudioFlinger::PlaybackThread::cacheParameters_l()
- {
- mSinkBufferSize = mNormalFrameCount * mFrameSize;
- activeSleepTime = activeSleepTimeUs();
- idleSleepTime = idleSleepTimeUs();
- }
- void AudioFlinger::PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType)
- {
- size_t size = mTracks.size();
- for (size_t i = 0; i < size; i++) {
- sp<Track> t = mTracks[i];
- if (t->streamType() == streamType) {
- t->invalidate();
- }
- }
- }
- void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
- {
- Mutex::Autolock _l(mLock);
- ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
- this, streamType, mTracks.size());
- invalidateTracks_l(streamType);
- }
- void AudioFlinger::PlaybackThread::onFatalError()
- {
- invalidateTracks(AUDIO_STREAM_MUSIC);
- }
- status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
- {
- int session = chain->sessionId();
- int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled
- ? mEffectBuffer : mSinkBuffer);
- bool ownsBuffer = false;
- ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
- if (session > 0) {
- // Only one effect chain can be present in direct output thread and it uses
- // the sink buffer as input
- if (mType != DIRECT) {
- size_t numSamples = mNormalFrameCount * mChannelCount;
- buffer = new int16_t[numSamples];
- memset(buffer, 0, numSamples * sizeof(int16_t));
- ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
- ownsBuffer = true;
- }
- // Attach all tracks with same session ID to this chain.
- for (size_t i = 0; i < mTracks.size(); ++i) {
- sp<Track> track = mTracks[i];
- if (session == track->sessionId()) {
- ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(),
- buffer);
- track->setMainBuffer(buffer);
- chain->incTrackCnt();
- }
- }
- // indicate all active tracks in the chain
- for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
- sp<Track> track = mActiveTracks[i].promote();
- if (track == 0) {
- continue;
- }
- if (session == track->sessionId()) {
- ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
- chain->incActiveTrackCnt();
- }
- }
- }
- chain->setThread(this);
- chain->setInBuffer(buffer, ownsBuffer);
- chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled
- ? mEffectBuffer : mSinkBuffer));
- // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
- // chains list in order to be processed last as it contains output stage effects
- // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
- // session AUDIO_SESSION_OUTPUT_STAGE to be processed
- // after track specific effects and before output stage
- // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
- // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
- // Effect chain for other sessions are inserted at beginning of effect
- // chains list to be processed before output mix effects. Relative order between other
- // sessions is not important
- size_t size = mEffectChains.size();
- size_t i = 0;
- for (i = 0; i < size; i++) {
- if (mEffectChains[i]->sessionId() < session) {
- break;
- }
- }
- mEffectChains.insertAt(chain, i);
- checkSuspendOnAddEffectChain_l(chain);
- return NO_ERROR;
- }
- size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
- {
- int session = chain->sessionId();
- ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
- for (size_t i = 0; i < mEffectChains.size(); i++) {
- if (chain == mEffectChains[i]) {
- mEffectChains.removeAt(i);
- // detach all active tracks from the chain
- for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
- sp<Track> track = mActiveTracks[i].promote();
- if (track == 0) {
- continue;
- }
- if (session == track->sessionId()) {
- ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
- chain.get(), session);
- chain->decActiveTrackCnt();
- }
- }
- // detach all tracks with same session ID from this chain
- for (size_t i = 0; i < mTracks.size(); ++i) {
- sp<Track> track = mTracks[i];
- if (session == track->sessionId()) {
- track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer));
- chain->decTrackCnt();
- }
- }
- break;
- }
- }
- return mEffectChains.size();
- }
- status_t AudioFlinger::PlaybackThread::attachAuxEffect(
- const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
- {
- Mutex::Autolock _l(mLock);
- return attachAuxEffect_l(track, EffectId);
- }
- status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
- const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
- {
- status_t status = NO_ERROR;
- if (EffectId == 0) {
- track->setAuxBuffer(0, NULL);
- } else {
- // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
- sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
- if (effect != 0) {
- if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
- track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
- } else {
- status = INVALID_OPERATION;
- }
- } else {
- status = BAD_VALUE;
- }
- }
- return status;
- }
- void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
- {
- for (size_t i = 0; i < mTracks.size(); ++i) {
- sp<Track> track = mTracks[i];
- if (track->auxEffectId() == effectId) {
- attachAuxEffect_l(track, 0);
- }
- }
- }
- bool AudioFlinger::PlaybackThread::threadLoop()
- {
- Vector< sp<Track> > tracksToRemove;
- standbyTime = systemTime();
- // MIXER
- nsecs_t lastWarning = 0;
- // DUPLICATING
- // FIXME could this be made local to while loop?
- writeFrames = 0;
- int lastGeneration = 0;
- cacheParameters_l();
- sleepTime = idleSleepTime;
- if (mType == MIXER) {
- sleepTimeShift = 0;
- }
- CpuStats cpuStats;
- const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid()));
- acquireWakeLock();
- // mNBLogWriter->log can only be called while thread mutex mLock is held.
- // So if you need to log when mutex is unlocked, set logString to a non-NULL string,
- // and then that string will be logged at the next convenient opportunity.
- const char *logString = NULL;
- checkSilentMode_l();
- while (!exitPending())
- {
- cpuStats.sample(myName);
- Vector< sp<EffectChain> > effectChains;
- { // scope for mLock
- Mutex::Autolock _l(mLock);
- processConfigEvents_l();
- if (logString != NULL) {
- mNBLogWriter->logTimestamp();
- mNBLogWriter->log(logString);
- logString = NULL;
- }
- // Gather the framesReleased counters for all active tracks,
- // and latch them atomically with the timestamp.
- // FIXME We're using raw pointers as indices. A unique track ID would be a better index.
- mLatchD.mFramesReleased.clear();
- size_t size = mActiveTracks.size();
- for (size_t i = 0; i < size; i++) {
- sp<Track> t = mActiveTracks[i].promote();
- if (t != 0) {
- mLatchD.mFramesReleased.add(t.get(),
- t->mAudioTrackServerProxy->framesReleased());
- }
- }
- if (mLatchDValid) {
- mLatchQ = mLatchD;
- mLatchDValid = false;
- mLatchQValid = true;
- }
- saveOutputTracks();
- if (mSignalPending) {
- // A signal was raised while we were unlocked
- mSignalPending = false;
- } else if (waitingAsyncCallback_l()) {
- if (exitPending()) {
- break;
- }
- releaseWakeLock_l();
- mWakeLockUids.clear();
- mActiveTracksGeneration++;
- ALOGV("wait async completion");
- mWaitWorkCV.wait(mLock);
- ALOGV("async completion/wake");
- acquireWakeLock_l();
- standbyTime = systemTime() + standbyDelay;
- sleepTime = 0;
- continue;
- }
- if ((!mActiveTracks.size() && systemTime() > standbyTime) ||
- isSuspended()) {
- // put audio hardware into standby after short delay
- if (shouldStandby_l()) {
- threadLoop_standby();
- mStandby = true;
- }
- if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
- // we're about to wait, flush the binder command buffer
- IPCThreadState::self()->flushCommands();
- clearOutputTracks();
- if (exitPending()) {
- break;
- }
- releaseWakeLock_l();
- mWakeLockUids.clear();
- mActiveTracksGeneration++;
- // wait until we have something to do...
- ALOGV("%s going to sleep", myName.string());
- mWaitWorkCV.wait(mLock);
- ALOGV("%s waking up", myName.string());
- acquireWakeLock_l();
- mMixerStatus = MIXER_IDLE;
- mMixerStatusIgnoringFastTracks = MIXER_IDLE;
- mBytesWritten = 0;
- mBytesRemaining = 0;
- checkSilentMode_l();
- standbyTime = systemTime() + standbyDelay;
- sleepTime = idleSleepTime;
- if (mType == MIXER) {
- sleepTimeShift = 0;
- }
- continue;
- }
- }
- // mMixerStatusIgnoringFastTracks is also updated internally
- mMixerStatus = prepareTracks_l(&tracksToRemove);
- // compare with previously applied list
- if (lastGeneration != mActiveTracksGeneration) {
- // update wakelock
- updateWakeLockUids_l(mWakeLockUids);
- lastGeneration = mActiveTracksGeneration;
- }
- // prevent any changes in effect chain list and in each effect chain
- // during mixing and effect process as the audio buffers could be deleted
- // or modified if an effect is created or deleted
- lockEffectChains_l(effectChains);
- } // mLock scope ends
- if (mBytesRemaining == 0) {
- mCurrentWriteLength = 0;
- if (mMixerStatus == MIXER_TRACKS_READY) {
- // threadLoop_mix() sets mCurrentWriteLength
- threadLoop_mix();
- } else if ((mMixerStatus != MIXER_DRAIN_TRACK)
- && (mMixerStatus != MIXER_DRAIN_ALL)) {
- // threadLoop_sleepTime sets sleepTime to 0 if data
- // must be written to HAL
- threadLoop_sleepTime();
- if (sleepTime == 0) {
- mCurrentWriteLength = mSinkBufferSize;
- }
- }
- // Either threadLoop_mix() or threadLoop_sleepTime() should have set
- // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0.
- // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid)
- // or mSinkBuffer (if there are no effects).
- //
- // This is done pre-effects computation; if effects change to
- // support higher precision, this needs to move.
- //
- // mMixerBufferValid is only set true by MixerThread::prepareTracks_l().
- // TODO use sleepTime == 0 as an additional condition.
- if (mMixerBufferValid) {
- void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer;
- audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat;
- memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat,
- mNormalFrameCount * mChannelCount);
- }
- mBytesRemaining = mCurrentWriteLength;
- if (isSuspended()) {
- sleepTime = suspendSleepTimeUs();
- // simulate write to HAL when suspended
- mBytesWritten += mSinkBufferSize;
- mBytesRemaining = 0;
- }
- // only process effects if we're going to write
- if (sleepTime == 0 && mType != OFFLOAD) {
- for (size_t i = 0; i < effectChains.size(); i ++) {
- #ifdef QCOM_DIRECTTRACK
- if (effectChains[i] != mAudioFlinger->mLPAEffectChain) {
- #endif
- effectChains[i]->process_l();
- #ifdef QCOM_DIRECTTRACK
- }
- #endif
- }
- }
- }
- // Process effect chains for offloaded thread even if no audio
- // was read from audio track: process only updates effect state
- // and thus does have to be synchronized with audio writes but may have
- // to be called while waiting for async write callback
- if (mType == OFFLOAD) {
- for (size_t i = 0; i < effectChains.size(); i ++) {
- effectChains[i]->process_l();
- }
- }
- // Only if the Effects buffer is enabled and there is data in the
- // Effects buffer (buffer valid), we need to
- // copy into the sink buffer.
- // TODO use sleepTime == 0 as an additional condition.
- if (mEffectBufferValid) {
- //ALOGV("writing effect buffer to sink buffer format %#x", mFormat);
- memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat,
- mNormalFrameCount * mChannelCount);
- }
- // enable changes in effect chain
- unlockEffectChains(effectChains);
- if (!waitingAsyncCallback()) {
- // sleepTime == 0 means we must write to audio hardware
- if (sleepTime == 0) {
- if (mBytesRemaining) {
- ssize_t ret = threadLoop_write();
- if (ret < 0) {
- mBytesRemaining = 0;
- #ifdef QCOM_DIRECTTRACK
- } else if(ret > mBytesRemaining) {
- mBytesWritten += mBytesRemaining;
- mBytesRemaining = 0;
- #endif
- } else {
- mBytesWritten += ret;
- mBytesRemaining -= ret;
- }
- } else if ((mMixerStatus == MIXER_DRAIN_TRACK) ||
- (mMixerStatus == MIXER_DRAIN_ALL)) {
- threadLoop_drain();
- }
- if (mType == MIXER) {
- // write blocked detection
- nsecs_t now = systemTime();
- nsecs_t delta = now - mLastWriteTime;
- if (!mStandby && delta > maxPeriod) {
- mNumDelayedWrites++;
- if ((now - lastWarning) > kWarningThrottleNs) {
- ATRACE_NAME("underrun");
- ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
- ns2ms(delta), mNumDelayedWrites, this);
- lastWarning = now;
- }
- }
- }
- } else {
- usleep(sleepTime);
- }
- }
- // Finally let go of removed track(s), without the lock held
- // since we can't guarantee the destructors won't acquire that
- // same lock. This will also mutate and push a new fast mixer state.
- threadLoop_removeTracks(tracksToRemove);
- tracksToRemove.clear();
- // FIXME I don't understand the need for this here;
- // it was in the original code but maybe the
- // assignment in saveOutputTracks() makes this unnecessary?
- clearOutputTracks();
- // Effect chains will be actually deleted here if they were removed from
- // mEffectChains list during mixing or effects processing
- effectChains.clear();
- // FIXME Note that the above .clear() is no longer necessary since effectChains
- // is now local to this block, but will keep it for now (at least until merge done).
- }
- threadLoop_exit();
- if (!mStandby) {
- threadLoop_standby();
- mStandby = true;
- }
- releaseWakeLock();
- mWakeLockUids.clear();
- mActiveTracksGeneration++;
- ALOGV("Thread %p type %d exiting", this, mType);
- return false;
- }
- // removeTracks_l() must be called with ThreadBase::mLock held
- void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove)
- {
- size_t count = tracksToRemove.size();
- if (count > 0) {
- for (size_t i=0 ; i<count ; i++) {
- const sp<Track>& track = tracksToRemove.itemAt(i);
- mActiveTracks.remove(track);
- mWakeLockUids.remove(track->uid());
- mActiveTracksGeneration++;
- ALOGV("removeTracks_l removing track on session %d", track->sessionId());
- sp<EffectChain> chain = getEffectChain_l(track->sessionId());
- if (chain != 0) {
- ALOGV("stopping track on chain %p for session Id: %d", chain.get(),
- track->sessionId());
- chain->decActiveTrackCnt();
- }
- if (track->isTerminated()) {
- removeTrack_l(track);
- }
- }
- }
- }
- status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp)
- {
- if (mNormalSink != 0) {
- return mNormalSink->getTimestamp(timestamp);
- }
- if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) {
- uint64_t position64;
- int ret = mOutput->stream->get_presentation_position(
- mOutput->stream, &position64, ×tamp.mTime);
- if (ret == 0) {
- timestamp.mPosition = (uint32_t)position64;
- return NO_ERROR;
- }
- }
- return INVALID_OPERATION;
- }
- status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch,
- audio_patch_handle_t *handle)
- {
- status_t status = NO_ERROR;
- if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
- // store new device and send to effects
- audio_devices_t type = AUDIO_DEVICE_NONE;
- for (unsigned int i = 0; i < patch->num_sinks; i++) {
- type |= patch->sinks[i].ext.device.type;
- }
- mOutDevice = type;
- for (size_t i = 0; i < mEffectChains.size(); i++) {
- mEffectChains[i]->setDevice_l(mOutDevice);
- }
- audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice();
- status = hwDevice->create_audio_patch(hwDevice,
- patch->num_sources,
- patch->sources,
- patch->num_sinks,
- patch->sinks,
- handle);
- } else {
- ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL");
- }
- return status;
- }
- status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
- {
- status_t status = NO_ERROR;
- if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
- audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice();
- status = hwDevice->release_audio_patch(hwDevice, handle);
- } else {
- ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL");
- }
- return status;
- }
- void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track)
- {
- Mutex::Autolock _l(mLock);
- mTracks.add(track);
- }
- void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track)
- {
- Mutex::Autolock _l(mLock);
- destroyTrack_l(track);
- }
- void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config)
- {
- ThreadBase::getAudioPortConfig(config);
- config->role = AUDIO_PORT_ROLE_SOURCE;
- config->ext.mix.hw_module = mOutput->audioHwDev->handle();
- config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
- }
- // ----------------------------------------------------------------------------
- AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, audio_devices_t device, type_t type)
- : PlaybackThread(audioFlinger, output, id, device, type),
- // mAudioMixer below
- // mFastMixer below
- mFastMixerFutex(0)
- // mOutputSink below
- // mPipeSink below
- // mNormalSink below
- {
- ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
- ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, "
- "mFrameCount=%d, mNormalFrameCount=%d",
- mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
- mNormalFrameCount);
- mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
- // create an NBAIO sink for the HAL output stream, and negotiate
- mOutputSink = new AudioStreamOutSink(output->stream);
- size_t numCounterOffers = 0;
- const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)};
- ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers);
- ALOG_ASSERT(index == 0);
- // initialize fast mixer depending on configuration
- bool initFastMixer;
- switch (kUseFastMixer) {
- case FastMixer_Never:
- initFastMixer = false;
- break;
- case FastMixer_Always:
- initFastMixer = true;
- break;
- case FastMixer_Static:
- case FastMixer_Dynamic:
- initFastMixer = mFrameCount < mNormalFrameCount;
- break;
- }
- if (initFastMixer) {
- audio_format_t fastMixerFormat;
- #ifndef QCOM_DIRECTTRACK
- if (mMixerBufferEnabled && mEffectBufferEnabled) {
- fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT;
- } else {
- #endif
- fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
- #ifndef QCOM_DIRECTTRACK
- }
- #endif
- if (mFormat != fastMixerFormat) {
- // change our Sink format to accept our intermediate precision
- mFormat = fastMixerFormat;
- free(mSinkBuffer);
- mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
- const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
- (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
- }
- // create a MonoPipe to connect our submix to FastMixer
- NBAIO_Format format = mOutputSink->format();
- NBAIO_Format origformat = format;
- // adjust format to match that of the Fast Mixer
- format.mFormat = fastMixerFormat;
- format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount;
- // This pipe depth compensates for scheduling latency of the normal mixer thread.
- // When it wakes up after a maximum latency, it runs a few cycles quickly before
- // finally blocking. Note the pipe implementation rounds up the request to a power of 2.
- MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/);
- const NBAIO_Format offers[1] = {format};
- size_t numCounterOffers = 0;
- ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers);
- ALOG_ASSERT(index == 0);
- monoPipe->setAvgFrames((mScreenState & 1) ?
- (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
- mPipeSink = monoPipe;
- #ifdef TEE_SINK
- if (mTeeSinkOutputEnabled) {
- // create a Pipe to archive a copy of FastMixer's output for dumpsys
- Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat);
- const NBAIO_Format offers2[1] = {origformat};
- numCounterOffers = 0;
- index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers);
- ALOG_ASSERT(index == 0);
- mTeeSink = teeSink;
- PipeReader *teeSource = new PipeReader(*teeSink);
- numCounterOffers = 0;
- index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers);
- ALOG_ASSERT(index == 0);
- mTeeSource = teeSource;
- }
- #endif
- // create fast mixer and configure it initially with just one fast track for our submix
- mFastMixer = new FastMixer();
- FastMixerStateQueue *sq = mFastMixer->sq();
- #ifdef STATE_QUEUE_DUMP
- sq->setObserverDump(&mStateQueueObserverDump);
- sq->setMutatorDump(&mStateQueueMutatorDump);
- #endif
- FastMixerState *state = sq->begin();
- FastTrack *fastTrack = &state->mFastTracks[0];
- // wrap the source side of the MonoPipe to make it an AudioBufferProvider
- fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe));
- fastTrack->mVolumeProvider = NULL;
- fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer
- fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer
- fastTrack->mGeneration++;
- state->mFastTracksGen++;
- state->mTrackMask = 1;
- // fast mixer will use the HAL output sink
- state->mOutputSink = mOutputSink.get();
- state->mOutputSinkGen++;
- state->mFrameCount = mFrameCount;
- state->mCommand = FastMixerState::COLD_IDLE;
- // already done in constructor initialization list
- //mFastMixerFutex = 0;
- state->mColdFutexAddr = &mFastMixerFutex;
- state->mColdGen++;
- state->mDumpState = &mFastMixerDumpState;
- #ifdef TEE_SINK
- state->mTeeSink = mTeeSink.get();
- #endif
- mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer");
- state->mNBLogWriter = mFastMixerNBLogWriter.get();
- sq->end();
- sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
- // start the fast mixer
- mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO);
- pid_t tid = mFastMixer->getTid();
- int err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
- if (err != 0) {
- ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
- kPriorityFastMixer, getpid_cached, tid, err);
- }
- #ifdef AUDIO_WATCHDOG
- // create and start the watchdog
- mAudioWatchdog = new AudioWatchdog();
- mAudioWatchdog->setDump(&mAudioWatchdogDump);
- mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO);
- tid = mAudioWatchdog->getTid();
- err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
- if (err != 0) {
- ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
- kPriorityFastMixer, getpid_cached, tid, err);
- }
- #endif
- }
- switch (kUseFastMixer) {
- case FastMixer_Never:
- case FastMixer_Dynamic:
- mNormalSink = mOutputSink;
- break;
- case FastMixer_Always:
- mNormalSink = mPipeSink;
- break;
- case FastMixer_Static:
- mNormalSink = initFastMixer ? mPipeSink : mOutputSink;
- break;
- }
- }
- AudioFlinger::MixerThread::~MixerThread()
- {
- if (mFastMixer != 0) {
- FastMixerStateQueue *sq = mFastMixer->sq();
- FastMixerState *state = sq->begin();
- if (state->mCommand == FastMixerState::COLD_IDLE) {
- int32_t old = android_atomic_inc(&mFastMixerFutex);
- if (old == -1) {
- (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
- }
- }
- state->mCommand = FastMixerState::EXIT;
- sq->end();
- sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
- mFastMixer->join();
- // Though the fast mixer thread has exited, it's state queue is still valid.
- // We'll use that extract the final state which contains one remaining fast track
- // corresponding to our sub-mix.
- state = sq->begin();
- ALOG_ASSERT(state->mTrackMask == 1);
- FastTrack *fastTrack = &state->mFastTracks[0];
- ALOG_ASSERT(fastTrack->mBufferProvider != NULL);
- delete fastTrack->mBufferProvider;
- sq->end(false /*didModify*/);
- mFastMixer.clear();
- #ifdef AUDIO_WATCHDOG
- if (mAudioWatchdog != 0) {
- mAudioWatchdog->requestExit();
- mAudioWatchdog->requestExitAndWait();
- mAudioWatchdog.clear();
- }
- #endif
- }
- mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter);
- delete mAudioMixer;
- }
- uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const
- {
- if (mFastMixer != 0) {
- MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
- latency += (pipe->getAvgFrames() * 1000) / mSampleRate;
- }
- return latency;
- }
- void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove)
- {
- PlaybackThread::threadLoop_removeTracks(tracksToRemove);
- }
- ssize_t AudioFlinger::MixerThread::threadLoop_write()
- {
- // FIXME we should only do one push per cycle; confirm this is true
- // Start the fast mixer if it's not already running
- if (mFastMixer != 0) {
- FastMixerStateQueue *sq = mFastMixer->sq();
- FastMixerState *state = sq->begin();
- if (state->mCommand != FastMixerState::MIX_WRITE &&
- (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) {
- if (state->mCommand == FastMixerState::COLD_IDLE) {
- int32_t old = android_atomic_inc(&mFastMixerFutex);
- if (old == -1) {
- (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
- }
- #ifdef AUDIO_WATCHDOG
- if (mAudioWatchdog != 0) {
- mAudioWatchdog->resume();
- }
- #endif
- }
- state->mCommand = FastMixerState::MIX_WRITE;
- mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
- FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN);
- sq->end();
- sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
- if (kUseFastMixer == FastMixer_Dynamic) {
- mNormalSink = mPipeSink;
- }
- } else {
- sq->end(false /*didModify*/);
- }
- }
- return PlaybackThread::threadLoop_write();
- }
- void AudioFlinger::MixerThread::threadLoop_standby()
- {
- // Idle the fast mixer if it's currently running
- if (mFastMixer != 0) {
- FastMixerStateQueue *sq = mFastMixer->sq();
- FastMixerState *state = sq->begin();
- if (!(state->mCommand & FastMixerState::IDLE)) {
- state->mCommand = FastMixerState::COLD_IDLE;
- state->mColdFutexAddr = &mFastMixerFutex;
- state->mColdGen++;
- mFastMixerFutex = 0;
- sq->end();
- // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
- sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED);
- if (kUseFastMixer == FastMixer_Dynamic) {
- mNormalSink = mOutputSink;
- }
- #ifdef AUDIO_WATCHDOG
- if (mAudioWatchdog != 0) {
- mAudioWatchdog->pause();
- }
- #endif
- } else {
- sq->end(false /*didModify*/);
- }
- }
- PlaybackThread::threadLoop_standby();
- }
- bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l()
- {
- return false;
- }
- bool AudioFlinger::PlaybackThread::shouldStandby_l()
- {
- return !mStandby;
- }
- bool AudioFlinger::PlaybackThread::waitingAsyncCallback()
- {
- Mutex::Autolock _l(mLock);
- return waitingAsyncCallback_l();
- }
- // shared by MIXER and DIRECT, overridden by DUPLICATING
- void AudioFlinger::PlaybackThread::threadLoop_standby()
- {
- ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended);
- mOutput->stream->common.standby(&mOutput->stream->common);
- if (mUseAsyncWrite != 0) {
- // discard any pending drain or write ack by incrementing sequence
- mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
- mDrainSequence = (mDrainSequence + 2) & ~1;
- ALOG_ASSERT(mCallbackThread != 0);
- mCallbackThread->setWriteBlocked(mWriteAckSequence);
- mCallbackThread->setDraining(mDrainSequence);
- }
- }
- void AudioFlinger::PlaybackThread::onAddNewTrack_l()
- {
- ALOGV("signal playback thread");
- broadcast_l();
- }
- void AudioFlinger::MixerThread::threadLoop_mix()
- {
- // obtain the presentation timestamp of the next output buffer
- int64_t pts;
- status_t status = INVALID_OPERATION;
- if (mNormalSink != 0) {
- status = mNormalSink->getNextWriteTimestamp(&pts);
- } else {
- status = mOutputSink->getNextWriteTimestamp(&pts);
- }
- if (status != NO_ERROR) {
- pts = AudioBufferProvider::kInvalidPTS;
- }
- // mix buffers...
- mAudioMixer->process(pts);
- mCurrentWriteLength = mSinkBufferSize;
- // increase sleep time progressively when application underrun condition clears.
- // Only increase sleep time if the mixer is ready for two consecutive times to avoid
- // that a steady state of alternating ready/not ready conditions keeps the sleep time
- // such that we would underrun the audio HAL.
- if ((sleepTime == 0) && (sleepTimeShift > 0)) {
- sleepTimeShift--;
- }
- sleepTime = 0;
- standbyTime = systemTime() + standbyDelay;
- //TODO: delay standby when effects have a tail
- }
- void AudioFlinger::MixerThread::threadLoop_sleepTime()
- {
- // If no tracks are ready, sleep once for the duration of an output
- // buffer size, then write 0s to the output
- if (sleepTime == 0) {
- if (mMixerStatus == MIXER_TRACKS_ENABLED) {
- sleepTime = activeSleepTime >> sleepTimeShift;
- if (sleepTime < kMinThreadSleepTimeUs) {
- sleepTime = kMinThreadSleepTimeUs;
- }
- // reduce sleep time in case of consecutive application underruns to avoid
- // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
- // duration we would end up writing less data than needed by the audio HAL if
- // the condition persists.
- if (sleepTimeShift < kMaxThreadSleepTimeShift) {
- sleepTimeShift++;
- }
- } else {
- sleepTime = idleSleepTime;
- }
- } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {
- // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared
- // before effects processing or output.
- if (mMixerBufferValid) {
- memset(mMixerBuffer, 0, mMixerBufferSize);
- } else {
- memset(mSinkBuffer, 0, mSinkBufferSize);
- }
- sleepTime = 0;
- ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED),
- "anticipated start");
- }
- // TODO add standby time extension fct of effect tail
- }
- // prepareTracks_l() must be called with ThreadBase::mLock held
- AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
- Vector< sp<Track> > *tracksToRemove)
- {
- mixer_state mixerStatus = MIXER_IDLE;
- // find out which tracks need to be processed
- size_t count = mActiveTracks.size();
- size_t mixedTracks = 0;
- size_t tracksWithEffect = 0;
- // counts only _active_ fast tracks
- size_t fastTracks = 0;
- uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset
- float masterVolume = mMasterVolume;
- bool masterMute = mMasterMute;
- if (masterMute) {
- masterVolume = 0;
- }
- // Delegate master volume control to effect in output mix effect chain if needed
- sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
- if (chain != 0) {
- uint32_t v = (uint32_t)(masterVolume * (1 << 24));
- chain->setVolume_l(&v, &v);
- masterVolume = (float)((v + (1 << 23)) >> 24);
- chain.clear();
- }
- // prepare a new state to push
- FastMixerStateQueue *sq = NULL;
- FastMixerState *state = NULL;
- bool didModify = false;
- FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED;
- if (mFastMixer != 0) {
- sq = mFastMixer->sq();
- state = sq->begin();
- }
- mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found.
- mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found.
- for (size_t i=0 ; i<count ; i++) {
- const sp<Track> t = mActiveTracks[i].promote();
- if (t == 0) {
- continue;
- }
- // this const just means the local variable doesn't change
- Track* const track = t.get();
- // process fast tracks
- if (track->isFastTrack()) {
- // It's theoretically possible (though unlikely) for a fast track to be created
- // and then removed within the same normal mix cycle. This is not a problem, as
- // the track never becomes active so it's fast mixer slot is never touched.
- // The converse, of removing an (active) track and then creating a new track
- // at the identical fast mixer slot within the same normal mix cycle,
- // is impossible because the slot isn't marked available until the end of each cycle.
- int j = track->mFastIndex;
- ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks);
- ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j)));
- FastTrack *fastTrack = &state->mFastTracks[j];
- // Determine whether the track is currently in underrun condition,
- // and whether it had a recent underrun.
- FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j];
- FastTrackUnderruns underruns = ftDump->mUnderruns;
- uint32_t recentFull = (underruns.mBitFields.mFull -
- track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK;
- uint32_t recentPartial = (underruns.mBitFields.mPartial -
- track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK;
- uint32_t recentEmpty = (underruns.mBitFields.mEmpty -
- track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK;
- uint32_t recentUnderruns = recentPartial + recentEmpty;
- track->mObservedUnderruns = underruns;
- // don't count underruns that occur while stopping or pausing
- // or stopped which can occur when flush() is called while active
- if (!(track->isStopping() || track->isPausing() || track->isStopped()) &&
- recentUnderruns > 0) {
- // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun
- track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount);
- }
- // This is similar to the state machine for normal tracks,
- // with a few modifications for fast tracks.
- bool isActive = true;
- switch (track->mState) {
- case TrackBase::STOPPING_1:
- // track stays active in STOPPING_1 state until first underrun
- if (recentUnderruns > 0 || track->isTerminated()) {
- track->mState = TrackBase::STOPPING_2;
- }
- break;
- case TrackBase::PAUSING:
- // ramp down is not yet implemented
- track->setPaused();
- break;
- case TrackBase::RESUMING:
- // ramp up is not yet implemented
- track->mState = TrackBase::ACTIVE;
- break;
- case TrackBase::ACTIVE:
- if (recentFull > 0 || recentPartial > 0) {
- // track has provided at least some frames recently: reset retry count
- track->mRetryCount = kMaxTrackRetries;
- }
- if (recentUnderruns == 0) {
- // no recent underruns: stay active
- break;
- }
- // there has recently been an underrun of some kind
- if (track->sharedBuffer() == 0) {
- // were any of the recent underruns "empty" (no frames available)?
- if (recentEmpty == 0) {
- // no, then ignore the partial underruns as they are allowed indefinitely
- break;
- }
- // there has recently been an "empty" underrun: decrement the retry counter
- if (--(track->mRetryCount) > 0) {
- break;
- }
- // indicate to client process that the track was disabled because of underrun;
- // it will then automatically call start() when data is available
- android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags);
- // remove from active list, but state remains ACTIVE [confusing but true]
- isActive = false;
- break;
- }
- else {
- if (recentEmpty == 0) {
- // no, then ignore the partial underruns as they are allowed indefinitely
- break;
- }
- }
- // fall through
- case TrackBase::STOPPING_2:
- case TrackBase::PAUSED:
- case TrackBase::STOPPED:
- case TrackBase::FLUSHED: // flush() while active
- // Check for presentation complete if track is inactive
- // We have consumed all the buffers of this track.
- // This would be incomplete if we auto-paused on underrun
- {
- size_t audioHALFrames =
- (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
- size_t framesWritten = mBytesWritten / mFrameSize;
- if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) {
- // track stays in active list until presentation is complete
- break;
- }
- }
- if (track->isStopping_2()) {
- track->mState = TrackBase::STOPPED;
- }
- if (track->isStopped()) {
- // Can't reset directly, as fast mixer is still polling this track
- // track->reset();
- // So instead mark this track as needing to be reset after push with ack
- resetMask |= 1 << i;
- }
- isActive = false;
- break;
- case TrackBase::IDLE:
- default:
- LOG_ALWAYS_FATAL("unexpected track state %d", track->mState);
- }
- if (isActive) {
- // was it previously inactive?
- if (!(state->mTrackMask & (1 << j))) {
- ExtendedAudioBufferProvider *eabp = track;
- VolumeProvider *vp = track;
- fastTrack->mBufferProvider = eabp;
- fastTrack->mVolumeProvider = vp;
- fastTrack->mChannelMask = track->mChannelMask;
- fastTrack->mFormat = track->mFormat;
- fastTrack->mGeneration++;
- state->mTrackMask |= 1 << j;
- didModify = true;
- // no acknowledgement required for newly active tracks
- }
- // cache the combined master volume and stream type volume for fast mixer; this
- // lacks any synchronization or barrier so VolumeProvider may read a stale value
- track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume;
- ++fastTracks;
- } else {
- // was it previously active?
- if (state->mTrackMask & (1 << j)) {
- fastTrack->mBufferProvider = NULL;
- fastTrack->mGeneration++;
- state->mTrackMask &= ~(1 << j);
- didModify = true;
- // If any fast tracks were removed, we must wait for acknowledgement
- // because we're about to decrement the last sp<> on those tracks.
- block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
- } else {
- LOG_ALWAYS_FATAL("fast track %d should have been active", j);
- }
- tracksToRemove->add(track);
- // Avoids a misleading display in dumpsys
- track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL;
- }
- continue;
- }
- { // local variable scope to avoid goto warning
- audio_track_cblk_t* cblk = track->cblk();
- // The first time a track is added we wait
- // for all its buffers to be filled before processing it
- int name = track->name();
- // make sure that we have enough frames to mix one full buffer.
- // enforce this condition only once to enable draining the buffer in case the client
- // app does not call stop() and relies on underrun to stop:
- // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
- // during last round
- size_t desiredFrames;
- uint32_t sr = track->sampleRate();
- if (sr == mSampleRate) {
- desiredFrames = mNormalFrameCount;
- } else {
- // +1 for rounding and +1 for additional sample needed for interpolation
- desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1;
- // add frames already consumed but not yet released by the resampler
- // because mAudioTrackServerProxy->framesReady() will include these frames
- desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
- #if 0
- // the minimum track buffer size is normally twice the number of frames necessary
- // to fill one buffer and the resampler should not leave more than one buffer worth
- // of unreleased frames after each pass, but just in case...
- ALOG_ASSERT(desiredFrames <= cblk->frameCount_);
- #endif
- }
- uint32_t minFrames = 1;
- if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
- (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
- minFrames = desiredFrames;
- }
- size_t framesReady = track->framesReady();
- if ((framesReady >= minFrames) && track->isReady() &&
- !track->isPaused() && !track->isTerminated())
- {
- ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this);
- mixedTracks++;
- // track->mainBuffer() != mSinkBuffer or mMixerBuffer means
- // there is an effect chain connected to the track
- chain.clear();
- if (track->mainBuffer() != mSinkBuffer &&
- track->mainBuffer() != mMixerBuffer) {
- if (mEffectBufferEnabled) {
- mEffectBufferValid = true; // Later can set directly.
- }
- chain = getEffectChain_l(track->sessionId());
- // Delegate volume control to effect in track effect chain if needed
- if (chain != 0) {
- tracksWithEffect++;
- } else {
- ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on "
- "session %d",
- name, track->sessionId());
- }
- }
- int param = AudioMixer::VOLUME;
- if (track->mFillingUpStatus == Track::FS_FILLED) {
- // no ramp for the first volume setting
- track->mFillingUpStatus = Track::FS_ACTIVE;
- if (track->mState == TrackBase::RESUMING) {
- track->mState = TrackBase::ACTIVE;
- param = AudioMixer::RAMP_VOLUME;
- }
- mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
- // FIXME should not make a decision based on mServer
- } else if (cblk->mServer != 0) {
- // If the track is stopped before the first frame was mixed,
- // do not apply ramp
- param = AudioMixer::RAMP_VOLUME;
- }
- // compute volume for this track
- uint32_t vl, vr; // in U8.24 integer format
- float vlf, vrf, vaf; // in [0.0, 1.0] float format
- if (track->isPausing() || mStreamTypes[track->streamType()].mute) {
- vl = vr = 0;
- vlf = vrf = vaf = 0.;
- if (track->isPausing()) {
- track->setPaused();
- }
- } else {
- // read original volumes with volume control
- float typeVolume = mStreamTypes[track->streamType()].volume;
- float v = masterVolume * typeVolume;
- AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
- gain_minifloat_packed_t vlr = proxy->getVolumeLR();
- vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
- vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
- // track volumes come from shared memory, so can't be trusted and must be clamped
- if (vlf > GAIN_FLOAT_UNITY) {
- ALOGV("Track left volume out of range: %.3g", vlf);
- vlf = GAIN_FLOAT_UNITY;
- }
- if (vrf > GAIN_FLOAT_UNITY) {
- ALOGV("Track right volume out of range: %.3g", vrf);
- vrf = GAIN_FLOAT_UNITY;
- }
- // now apply the master volume and stream type volume
- vlf *= v;
- vrf *= v;
- // assuming master volume and stream type volume each go up to 1.0,
- // then derive vl and vr as U8.24 versions for the effect chain
- const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT;
- vl = (uint32_t) (scaleto8_24 * vlf);
- vr = (uint32_t) (scaleto8_24 * vrf);
- // vl and vr are now in U8.24 format
- uint16_t sendLevel = proxy->getSendLevel_U4_12();
- // send level comes from shared memory and so may be corrupt
- if (sendLevel > MAX_GAIN_INT) {
- ALOGV("Track send level out of range: %04X", sendLevel);
- sendLevel = MAX_GAIN_INT;
- }
- // vaf is represented as [0.0, 1.0] float by rescaling sendLevel
- vaf = v * sendLevel * (1. / MAX_GAIN_INT);
- }
- // Delegate volume control to effect in track effect chain if needed
- if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
- // Do not ramp volume if volume is controlled by effect
- param = AudioMixer::VOLUME;
- // Update remaining floating point volume levels
- vlf = (float)vl / (1 << 24);
- vrf = (float)vr / (1 << 24);
- track->mHasVolumeController = true;
- } else {
- // force no volume ramp when volume controller was just disabled or removed
- // from effect chain to avoid volume spike
- if (track->mHasVolumeController) {
- param = AudioMixer::VOLUME;
- }
- track->mHasVolumeController = false;
- }
- // XXX: these things DON'T need to be done each time
- mAudioMixer->setBufferProvider(name, track);
- mAudioMixer->enable(name);
- mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf);
- mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf);
- mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf);
- mAudioMixer->setParameter(
- name,
- AudioMixer::TRACK,
- AudioMixer::FORMAT, (void *)track->format());
- mAudioMixer->setParameter(
- name,
- AudioMixer::TRACK,
- AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask());
- mAudioMixer->setParameter(
- name,
- AudioMixer::TRACK,
- AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask);
- uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX;
- uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate();
- if (reqSampleRate == 0) {
- reqSampleRate = mSampleRate;
- } else if (reqSampleRate > maxSampleRate) {
- reqSampleRate = maxSampleRate;
- }
- mAudioMixer->setParameter(
- name,
- AudioMixer::RESAMPLE,
- AudioMixer::SAMPLE_RATE,
- (void *)(uintptr_t)reqSampleRate);
- /*
- * Select the appropriate output buffer for the track.
- *
- * Tracks with effects go into their own effects chain buffer
- * and from there into either mEffectBuffer or mSinkBuffer.
- *
- * Other tracks can use mMixerBuffer for higher precision
- * channel accumulation. If this buffer is enabled
- * (mMixerBufferEnabled true), then selected tracks will accumulate
- * into it.
- *
- */
- if (mMixerBufferEnabled
- && (track->mainBuffer() == mSinkBuffer
- || track->mainBuffer() == mMixerBuffer)) {
- mAudioMixer->setParameter(
- name,
- AudioMixer::TRACK,
- AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
- mAudioMixer->setParameter(
- name,
- AudioMixer::TRACK,
- AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer);
- // TODO: override track->mainBuffer()?
- mMixerBufferValid = true;
- } else {
- mAudioMixer->setParameter(
- name,
- AudioMixer::TRACK,
- AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT);
- mAudioMixer->setParameter(
- name,
- AudioMixer::TRACK,
- AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
- }
- mAudioMixer->setParameter(
- name,
- AudioMixer::TRACK,
- AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
- // reset retry count
- track->mRetryCount = kMaxTrackRetries;
- // If one track is ready, set the mixer ready if:
- // - the mixer was not ready during previous round OR
- // - no other track is not ready
- if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY ||
- mixerStatus != MIXER_TRACKS_ENABLED) {
- mixerStatus = MIXER_TRACKS_READY;
- }
- } else {
- if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) {
- track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
- }
- // clear effect chain input buffer if an active track underruns to avoid sending
- // previous audio buffer again to effects
- chain = getEffectChain_l(track->sessionId());
- if (chain != 0) {
- chain->clearInputBuffer();
- }
- ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this);
- if ((track->sharedBuffer() != 0) || track->isTerminated() ||
- track->isStopped() || track->isPaused()) {
- // We have consumed all the buffers of this track.
- // Remove it from the list of active tracks.
- // TODO: use actual buffer filling status instead of latency when available from
- // audio HAL
- size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
- size_t framesWritten = mBytesWritten / mFrameSize;
- if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
- if (track->isStopped()) {
- track->reset();
- }
- tracksToRemove->add(track);
- }
- } else {
- // No buffers for this track. Give it a few chances to
- // fill a buffer, then remove it from active list.
- if (--(track->mRetryCount) <= 0) {
- ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
- tracksToRemove->add(track);
- // indicate to client process that the track was disabled because of underrun;
- // it will then automatically call start() when data is available
- android_atomic_or(CBLK_DISABLED, &cblk->mFlags);
- // If one track is not ready, mark the mixer also not ready if:
- // - the mixer was ready during previous round OR
- // - no other track is ready
- } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY ||
- mixerStatus != MIXER_TRACKS_READY) {
- mixerStatus = MIXER_TRACKS_ENABLED;
- }
- }
- mAudioMixer->disable(name);
- }
- } // local variable scope to avoid goto warning
- track_is_ready: ;
- }
- // Push the new FastMixer state if necessary
- bool pauseAudioWatchdog = false;
- if (didModify) {
- state->mFastTracksGen++;
- // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle
- if (kUseFastMixer == FastMixer_Dynamic &&
- state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) {
- state->mCommand = FastMixerState::COLD_IDLE;
- state->mColdFutexAddr = &mFastMixerFutex;
- state->mColdGen++;
- mFastMixerFutex = 0;
- if (kUseFastMixer == FastMixer_Dynamic) {
- mNormalSink = mOutputSink;
- }
- // If we go into cold idle, need to wait for acknowledgement
- // so that fast mixer stops doing I/O.
- block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
- pauseAudioWatchdog = true;
- }
- }
- if (sq != NULL) {
- sq->end(didModify);
- sq->push(block);
- }
- #ifdef AUDIO_WATCHDOG
- if (pauseAudioWatchdog && mAudioWatchdog != 0) {
- mAudioWatchdog->pause();
- }
- #endif
- // Now perform the deferred reset on fast tracks that have stopped
- while (resetMask != 0) {
- size_t i = __builtin_ctz(resetMask);
- ALOG_ASSERT(i < count);
- resetMask &= ~(1 << i);
- sp<Track> t = mActiveTracks[i].promote();
- if (t == 0) {
- continue;
- }
- Track* track = t.get();
- ALOG_ASSERT(track->isFastTrack() && track->isStopped());
- track->reset();
- }
- // remove all the tracks that need to be...
- removeTracks_l(*tracksToRemove);
- if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) {
- mEffectBufferValid = true;
- }
- if (mEffectBufferValid) {
- // as long as there are effects we should clear the effects buffer, to avoid
- // passing a non-clean buffer to the effect chain
- memset(mEffectBuffer, 0, mEffectBufferSize);
- }
- // sink or mix buffer must be cleared if all tracks are connected to an
- // effect chain as in this case the mixer will not write to the sink or mix buffer
- // and track effects will accumulate into it
- if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
- (mixedTracks == 0 && fastTracks > 0))) {
- // FIXME as a performance optimization, should remember previous zero status
- if (mMixerBufferValid) {
- memset(mMixerBuffer, 0, mMixerBufferSize);
- // TODO: In testing, mSinkBuffer below need not be cleared because
- // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer
- // after mixing.
- //
- // To enforce this guarantee:
- // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
- // (mixedTracks == 0 && fastTracks > 0))
- // must imply MIXER_TRACKS_READY.
- // Later, we may clear buffers regardless, and skip much of this logic.
- }
- // FIXME as a performance optimization, should remember previous zero status
- memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize);
- }
- // if any fast tracks, then status is ready
- mMixerStatusIgnoringFastTracks = mixerStatus;
- if (fastTracks > 0) {
- mixerStatus = MIXER_TRACKS_READY;
- }
- return mixerStatus;
- }
- // getTrackName_l() must be called with ThreadBase::mLock held
- int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask,
- audio_format_t format, int sessionId)
- {
- return mAudioMixer->getTrackName(channelMask, format, sessionId);
- }
- // deleteTrackName_l() must be called with ThreadBase::mLock held
- void AudioFlinger::MixerThread::deleteTrackName_l(int name)
- {
- ALOGV("remove track (%d) and delete from mixer", name);
- mAudioMixer->deleteTrackName(name);
- }
- // checkForNewParameter_l() must be called with ThreadBase::mLock held
- bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair,
- status_t& status)
- {
- bool reconfig = false;
- status = NO_ERROR;
- // if !&IDLE, holds the FastMixer state to restore after new parameters processed
- FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE;
- if (mFastMixer != 0) {
- FastMixerStateQueue *sq = mFastMixer->sq();
- FastMixerState *state = sq->begin();
- if (!(state->mCommand & FastMixerState::IDLE)) {
- previousCommand = state->mCommand;
- state->mCommand = FastMixerState::HOT_IDLE;
- sq->end();
- sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED);
- } else {
- sq->end(false /*didModify*/);
- }
- }
- AudioParameter param = AudioParameter(keyValuePair);
- int value;
- if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
- reconfig = true;
- }
- if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
- if (!isValidPcmSinkFormat((audio_format_t) value)) {
- status = BAD_VALUE;
- } else {
- // no need to save value, since it's constant
- reconfig = true;
- }
- }
- if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
- if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) {
- status = BAD_VALUE;
- } else {
- // no need to save value, since it's constant
- reconfig = true;
- }
- }
- if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
- // do not accept frame count changes if tracks are open as the track buffer
- // size depends on frame count and correct behavior would not be guaranteed
- // if frame count is changed after track creation
- if (!mTracks.isEmpty()) {
- status = INVALID_OPERATION;
- } else {
- reconfig = true;
- }
- }
- if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
- #ifdef ADD_BATTERY_DATA
- // when changing the audio output device, call addBatteryData to notify
- // the change
- if (mOutDevice != value) {
- uint32_t params = 0;
- // check whether speaker is on
- if (value & AUDIO_DEVICE_OUT_SPEAKER) {
- params |= IMediaPlayerService::kBatteryDataSpeakerOn;
- }
- audio_devices_t deviceWithoutSpeaker
- = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
- // check if any other device (except speaker) is on
- if (value & deviceWithoutSpeaker ) {
- params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
- }
- if (params != 0) {
- addBatteryData(params);
- }
- }
- #endif
- // forward device change to effects that have requested to be
- // aware of attached audio device.
- if (value != AUDIO_DEVICE_NONE) {
- mOutDevice = value;
- for (size_t i = 0; i < mEffectChains.size(); i++) {
- mEffectChains[i]->setDevice_l(mOutDevice);
- }
- }
- }
- if (status == NO_ERROR) {
- status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
- keyValuePair.string());
- if (!mStandby && status == INVALID_OPERATION) {
- mOutput->stream->common.standby(&mOutput->stream->common);
- mStandby = true;
- mBytesWritten = 0;
- status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
- keyValuePair.string());
- }
- if (status == NO_ERROR && reconfig) {
- readOutputParameters_l();
- delete mAudioMixer;
- mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
- for (size_t i = 0; i < mTracks.size() ; i++) {
- int name = getTrackName_l(mTracks[i]->mChannelMask,
- mTracks[i]->mFormat, mTracks[i]->mSessionId);
- if (name < 0) {
- break;
- }
- mTracks[i]->mName = name;
- }
- sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
- }
- }
- if (!(previousCommand & FastMixerState::IDLE)) {
- ALOG_ASSERT(mFastMixer != 0);
- FastMixerStateQueue *sq = mFastMixer->sq();
- FastMixerState *state = sq->begin();
- ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE);
- state->mCommand = previousCommand;
- sq->end();
- sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
- }
- return reconfig;
- }
- void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
- {
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
- PlaybackThread::dumpInternals(fd, args);
- dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames());
- // Make a non-atomic copy of fast mixer dump state so it won't change underneath us
- const FastMixerDumpState copy(mFastMixerDumpState);
- copy.dump(fd);
- #ifdef STATE_QUEUE_DUMP
- // Similar for state queue
- StateQueueObserverDump observerCopy = mStateQueueObserverDump;
- observerCopy.dump(fd);
- StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump;
- mutatorCopy.dump(fd);
- #endif
- #ifdef TEE_SINK
- // Write the tee output to a .wav file
- dumpTee(fd, mTeeSource, mId);
- #endif
- #ifdef AUDIO_WATCHDOG
- if (mAudioWatchdog != 0) {
- // Make a non-atomic copy of audio watchdog dump so it won't change underneath us
- AudioWatchdogDump wdCopy = mAudioWatchdogDump;
- wdCopy.dump(fd);
- }
- #endif
- }
- uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
- {
- return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2;
- }
- uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const
- {
- return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000);
- }
- void AudioFlinger::MixerThread::cacheParameters_l()
- {
- PlaybackThread::cacheParameters_l();
- // FIXME: Relaxed timing because of a certain device that can't meet latency
- // Should be reduced to 2x after the vendor fixes the driver issue
- // increase threshold again due to low power audio mode. The way this warning
- // threshold is calculated and its usefulness should be reconsidered anyway.
- maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15;
- }
- // ----------------------------------------------------------------------------
- AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
- AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device)
- : PlaybackThread(audioFlinger, output, id, device, DIRECT)
- // mLeftVolFloat, mRightVolFloat
- {
- }
- AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
- AudioStreamOut* output, audio_io_handle_t id, uint32_t device,
- ThreadBase::type_t type)
- : PlaybackThread(audioFlinger, output, id, device, type)
- // mLeftVolFloat, mRightVolFloat
- {
- }
- AudioFlinger::DirectOutputThread::~DirectOutputThread()
- {
- }
- void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack)
- {
- audio_track_cblk_t* cblk = track->cblk();
- float left, right;
- if (mMasterMute || mStreamTypes[track->streamType()].mute) {
- left = right = 0;
- } else {
- float typeVolume = mStreamTypes[track->streamType()].volume;
- float v = mMasterVolume * typeVolume;
- AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
- gain_minifloat_packed_t vlr = proxy->getVolumeLR();
- left = float_from_gain(gain_minifloat_unpack_left(vlr));
- if (left > GAIN_FLOAT_UNITY) {
- left = GAIN_FLOAT_UNITY;
- }
- left *= v;
- right = float_from_gain(gain_minifloat_unpack_right(vlr));
- if (right > GAIN_FLOAT_UNITY) {
- right = GAIN_FLOAT_UNITY;
- }
- right *= v;
- }
- if (lastTrack) {
- if (left != mLeftVolFloat || right != mRightVolFloat) {
- mLeftVolFloat = left;
- mRightVolFloat = right;
- // Convert volumes from float to 8.24
- uint32_t vl = (uint32_t)(left * (1 << 24));
- uint32_t vr = (uint32_t)(right * (1 << 24));
- // Delegate volume control to effect in track effect chain if needed
- // only one effect chain can be present on DirectOutputThread, so if
- // there is one, the track is connected to it
- if (!mEffectChains.isEmpty()) {
- mEffectChains[0]->setVolume_l(&vl, &vr);
- left = (float)vl / (1 << 24);
- right = (float)vr / (1 << 24);
- }
- if (mOutput->stream->set_volume) {
- mOutput->stream->set_volume(mOutput->stream, left, right);
- }
- }
- }
- }
- AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l(
- Vector< sp<Track> > *tracksToRemove
- )
- {
- size_t count = mActiveTracks.size();
- mixer_state mixerStatus = MIXER_IDLE;
- // find out which tracks need to be processed
- for (size_t i = 0; i < count; i++) {
- sp<Track> t = mActiveTracks[i].promote();
- // The track died recently
- if (t == 0) {
- continue;
- }
- Track* const track = t.get();
- audio_track_cblk_t* cblk = track->cblk();
- // Only consider last track started for volume and mixer state control.
- // In theory an older track could underrun and restart after the new one starts
- // but as we only care about the transition phase between two tracks on a
- // direct output, it is not a problem to ignore the underrun case.
- sp<Track> l = mLatestActiveTrack.promote();
- bool last = l.get() == track;
- // The first time a track is added we wait
- // for all its buffers to be filled before processing it
- uint32_t minFrames;
- if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) {
- minFrames = mNormalFrameCount;
- } else {
- minFrames = 1;
- }
- if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() &&
- !track->isStopping() && !track->isStopped())
- {
- ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer);
- if (track->mFillingUpStatus == Track::FS_FILLED) {
- track->mFillingUpStatus = Track::FS_ACTIVE;
- // make sure processVolume_l() will apply new volume even if 0
- mLeftVolFloat = mRightVolFloat = -1.0;
- if (track->mState == TrackBase::RESUMING) {
- track->mState = TrackBase::ACTIVE;
- }
- }
- // compute volume for this track
- processVolume_l(track, last);
- if (last) {
- // reset retry count
- track->mRetryCount = kMaxTrackRetriesDirect;
- mActiveTrack = t;
- mixerStatus = MIXER_TRACKS_READY;
- }
- } else {
- // clear effect chain input buffer if the last active track started underruns
- // to avoid sending previous audio buffer again to effects
- if (!mEffectChains.isEmpty() && last) {
- mEffectChains[0]->clearInputBuffer();
- }
- if (track->isStopping_1()) {
- track->mState = TrackBase::STOPPING_2;
- }
- if ((track->sharedBuffer() != 0) || track->isStopped() ||
- track->isStopping_2() || track->isPaused()) {
- // We have consumed all the buffers of this track.
- // Remove it from the list of active tracks.
- size_t audioHALFrames;
- if (audio_is_linear_pcm(mFormat)) {
- audioHALFrames = (latency_l() * mSampleRate) / 1000;
- } else {
- audioHALFrames = 0;
- }
- size_t framesWritten = mBytesWritten / mFrameSize;
- if (mStandby || !last ||
- track->presentationComplete(framesWritten, audioHALFrames)) {
- if (track->isStopping_2()) {
- track->mState = TrackBase::STOPPED;
- }
- if (track->isStopped()) {
- if (track->mState == TrackBase::FLUSHED) {
- flushHw_l();
- }
- track->reset();
- }
- tracksToRemove->add(track);
- }
- } else {
- // No buffers for this track. Give it a few chances to
- // fill a buffer, then remove it from active list.
- // Only consider last track started for mixer state control
- if (--(track->mRetryCount) <= 0) {
- ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
- tracksToRemove->add(track);
- // indicate to client process that the track was disabled because of underrun;
- // it will then automatically call start() when data is available
- android_atomic_or(CBLK_DISABLED, &cblk->mFlags);
- } else if (last) {
- mixerStatus = MIXER_TRACKS_ENABLED;
- }
- }
- }
- }
- // remove all the tracks that need to be...
- removeTracks_l(*tracksToRemove);
- return mixerStatus;
- }
- void AudioFlinger::DirectOutputThread::threadLoop_mix()
- {
- size_t frameCount = mFrameCount;
- int8_t *curBuf = (int8_t *)mSinkBuffer;
- // output audio to hardware
- while (frameCount) {
- AudioBufferProvider::Buffer buffer;
- buffer.frameCount = frameCount;
- mActiveTrack->getNextBuffer(&buffer);
- if (buffer.raw == NULL) {
- memset(curBuf, 0, frameCount * mFrameSize);
- break;
- }
- memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
- frameCount -= buffer.frameCount;
- curBuf += buffer.frameCount * mFrameSize;
- mActiveTrack->releaseBuffer(&buffer);
- }
- mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer;
- sleepTime = 0;
- standbyTime = systemTime() + standbyDelay;
- mActiveTrack.clear();
- }
- void AudioFlinger::DirectOutputThread::threadLoop_sleepTime()
- {
- if (sleepTime == 0) {
- if (mMixerStatus == MIXER_TRACKS_ENABLED) {
- sleepTime = activeSleepTime;
- } else {
- sleepTime = idleSleepTime;
- }
- } else if (mBytesWritten != 0 && (audio_is_linear_pcm(mFormat) ||
- audio_is_compress_voip_format(mFormat) ||
- audio_is_compress_capture_format(mFormat))) {
- memset(mSinkBuffer, 0, mFrameCount * mFrameSize);
- sleepTime = 0;
- }
- }
- // getTrackName_l() must be called with ThreadBase::mLock held
- int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused,
- audio_format_t format __unused, int sessionId __unused)
- {
- return 0;
- }
- // deleteTrackName_l() must be called with ThreadBase::mLock held
- void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused)
- {
- }
- // checkForNewParameter_l() must be called with ThreadBase::mLock held
- bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair,
- status_t& status)
- {
- bool reconfig = false;
- status = NO_ERROR;
- AudioParameter param = AudioParameter(keyValuePair);
- int value;
- if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
- // forward device change to effects that have requested to be
- // aware of attached audio device.
- if (value != AUDIO_DEVICE_NONE) {
- mOutDevice = value;
- for (size_t i = 0; i < mEffectChains.size(); i++) {
- mEffectChains[i]->setDevice_l(mOutDevice);
- }
- }
- }
- if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
- // do not accept frame count changes if tracks are open as the track buffer
- // size depends on frame count and correct behavior would not be garantied
- // if frame count is changed after track creation
- if (!mTracks.isEmpty()) {
- status = INVALID_OPERATION;
- } else {
- reconfig = true;
- }
- }
- if (status == NO_ERROR) {
- status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
- keyValuePair.string());
- if (!mStandby && status == INVALID_OPERATION) {
- mOutput->stream->common.standby(&mOutput->stream->common);
- mStandby = true;
- mBytesWritten = 0;
- status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
- keyValuePair.string());
- }
- if (status == NO_ERROR && reconfig) {
- readOutputParameters_l();
- sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
- }
- }
- return reconfig;
- }
- uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const
- {
- uint32_t time;
- if (audio_is_linear_pcm(mFormat)) {
- time = PlaybackThread::activeSleepTimeUs();
- } else {
- time = 10000;
- }
- return time;
- }
- uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const
- {
- uint32_t time;
- if (audio_is_linear_pcm(mFormat)) {
- time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
- } else {
- time = 10000;
- }
- return time;
- }
- uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const
- {
- uint32_t time;
- if (audio_is_linear_pcm(mFormat)) {
- time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
- } else {
- time = 10000;
- }
- return time;
- }
- void AudioFlinger::DirectOutputThread::cacheParameters_l()
- {
- PlaybackThread::cacheParameters_l();
- // use shorter standby delay as on normal output to release
- // hardware resources as soon as possible
- if (audio_is_linear_pcm(mFormat)) {
- standbyDelay = microseconds(activeSleepTime*2);
- } else {
- standbyDelay = kOffloadStandbyDelayNs;
- }
- }
- void AudioFlinger::DirectOutputThread::flushHw_l()
- {
- if (mOutput->stream->flush != NULL)
- mOutput->stream->flush(mOutput->stream);
- }
- // ----------------------------------------------------------------------------
- AudioFlinger::AsyncCallbackThread::AsyncCallbackThread(
- const wp<AudioFlinger::PlaybackThread>& playbackThread)
- : Thread(false /*canCallJava*/),
- mPlaybackThread(playbackThread),
- mWriteAckSequence(0),
- mDrainSequence(0)
- {
- }
- AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread()
- {
- }
- void AudioFlinger::AsyncCallbackThread::onFirstRef()
- {
- run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO);
- }
- bool AudioFlinger::AsyncCallbackThread::threadLoop()
- {
- while (!exitPending()) {
- uint32_t writeAckSequence;
- uint32_t drainSequence;
- {
- Mutex::Autolock _l(mLock);
- while (!((mWriteAckSequence & 1) ||
- (mDrainSequence & 1) ||
- exitPending())) {
- mWaitWorkCV.wait(mLock);
- }
- if (exitPending()) {
- break;
- }
- ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d",
- mWriteAckSequence, mDrainSequence);
- writeAckSequence = mWriteAckSequence;
- mWriteAckSequence &= ~1;
- drainSequence = mDrainSequence;
- mDrainSequence &= ~1;
- }
- {
- sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote();
- if (playbackThread != 0) {
- if (writeAckSequence & 1) {
- playbackThread->resetWriteBlocked(writeAckSequence >> 1);
- }
- if (drainSequence & 1) {
- playbackThread->resetDraining(drainSequence >> 1);
- }
- }
- }
- }
- return false;
- }
- void AudioFlinger::AsyncCallbackThread::exit()
- {
- ALOGV("AsyncCallbackThread::exit");
- Mutex::Autolock _l(mLock);
- requestExit();
- mWaitWorkCV.broadcast();
- }
- void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence)
- {
- Mutex::Autolock _l(mLock);
- // bit 0 is cleared
- mWriteAckSequence = sequence << 1;
- }
- void AudioFlinger::AsyncCallbackThread::resetWriteBlocked()
- {
- Mutex::Autolock _l(mLock);
- // ignore unexpected callbacks
- if (mWriteAckSequence & 2) {
- mWriteAckSequence |= 1;
- mWaitWorkCV.signal();
- }
- }
- void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence)
- {
- Mutex::Autolock _l(mLock);
- // bit 0 is cleared
- mDrainSequence = sequence << 1;
- }
- void AudioFlinger::AsyncCallbackThread::resetDraining()
- {
- Mutex::Autolock _l(mLock);
- // ignore unexpected callbacks
- if (mDrainSequence & 2) {
- mDrainSequence |= 1;
- mWaitWorkCV.signal();
- }
- }
- // ----------------------------------------------------------------------------
- AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger,
- AudioStreamOut* output, audio_io_handle_t id, uint32_t device)
- : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD),
- mHwPaused(false),
- mFlushPending(false),
- mPausedBytesRemaining(0)
- {
- //FIXME: mStandby should be set to true by ThreadBase constructor
- mStandby = true;
- }
- void AudioFlinger::OffloadThread::threadLoop_exit()
- {
- if (mFlushPending || mHwPaused) {
- // If a flush is pending or track was paused, just discard buffered data
- flushHw_l();
- } else {
- mMixerStatus = MIXER_DRAIN_ALL;
- threadLoop_drain();
- }
- if (mUseAsyncWrite) {
- ALOG_ASSERT(mCallbackThread != 0);
- mCallbackThread->exit();
- }
- PlaybackThread::threadLoop_exit();
- }
- AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l(
- Vector< sp<Track> > *tracksToRemove
- )
- {
- size_t count = mActiveTracks.size();
- mixer_state mixerStatus = MIXER_IDLE;
- bool doHwPause = false;
- bool doHwResume = false;
- ALOGV("OffloadThread::prepareTracks_l active tracks %d", count);
- // find out which tracks need to be processed
- for (size_t i = 0; i < count; i++) {
- sp<Track> t = mActiveTracks[i].promote();
- // The track died recently
- if (t == 0) {
- continue;
- }
- Track* const track = t.get();
- audio_track_cblk_t* cblk = track->cblk();
- // Only consider last track started for volume and mixer state control.
- // In theory an older track could underrun and restart after the new one starts
- // but as we only care about the transition phase between two tracks on a
- // direct output, it is not a problem to ignore the underrun case.
- sp<Track> l = mLatestActiveTrack.promote();
- bool last = l.get() == track;
- if (track->isInvalid()) {
- ALOGW("An invalidated track shouldn't be in active list");
- tracksToRemove->add(track);
- continue;
- }
- if (track->mState == TrackBase::IDLE) {
- ALOGW("An idle track shouldn't be in active list");
- continue;
- }
- if (track->isPausing()) {
- track->setPaused();
- if (last) {
- if (!mHwPaused) {
- doHwPause = true;
- mHwPaused = true;
- }
- // If we were part way through writing the mixbuffer to
- // the HAL we must save this until we resume
- // BUG - this will be wrong if a different track is made active,
- // in that case we want to discard the pending data in the
- // mixbuffer and tell the client to present it again when the
- // track is resumed
- mPausedWriteLength = mCurrentWriteLength;
- mPausedBytesRemaining = mBytesRemaining;
- mBytesRemaining = 0; // stop writing
- }
- tracksToRemove->add(track);
- } else if (track->isFlushPending()) {
- track->flushAck();
- if (last) {
- mFlushPending = true;
- }
- } else if (track->isResumePending()){
- track->resumeAck();
- if (last) {
- if (mPausedBytesRemaining) {
- // Need to continue write that was interrupted
- mCurrentWriteLength = mPausedWriteLength;
- mBytesRemaining = mPausedBytesRemaining;
- mPausedBytesRemaining = 0;
- }
- if (mHwPaused) {
- doHwResume = true;
- mHwPaused = false;
- // threadLoop_mix() will handle the case that we need to
- // resume an interrupted write
- }
- // enable write to audio HAL
- sleepTime = 0;
- // Do not handle new data in this iteration even if track->framesReady()
- mixerStatus = MIXER_TRACKS_ENABLED;
- }
- } else if (track->framesReady() && track->isReady() &&
- !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) {
- ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer);
- if (track->mFillingUpStatus == Track::FS_FILLED) {
- track->mFillingUpStatus = Track::FS_ACTIVE;
- // make sure processVolume_l() will apply new volume even if 0
- mLeftVolFloat = mRightVolFloat = -1.0;
- }
- if (last) {
- sp<Track> previousTrack = mPreviousTrack.promote();
- if (previousTrack != 0) {
- if (track != previousTrack.get()) {
- // Flush any data still being written from last track
- mBytesRemaining = 0;
- if (mPausedBytesRemaining) {
- // Last track was paused so we also need to flush saved
- // mixbuffer state and invalidate track so that it will
- // re-submit that unwritten data when it is next resumed
- mPausedBytesRemaining = 0;
- // Invalidate is a bit drastic - would be more efficient
- // to have a flag to tell client that some of the
- // previously written data was lost
- previousTrack->invalidate();
- }
- // flush data already sent to the DSP if changing audio session as audio
- // comes from a different source. Also invalidate previous track to force a
- // seek when resuming.
- if (previousTrack->sessionId() != track->sessionId()) {
- previousTrack->invalidate();
- }
- }
- }
- mPreviousTrack = track;
- // reset retry count
- track->mRetryCount = kMaxTrackRetriesOffload;
- mActiveTrack = t;
- mixerStatus = MIXER_TRACKS_READY;
- }
- } else {
- ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer);
- if (track->isStopping_1()) {
- // Hardware buffer can hold a large amount of audio so we must
- // wait for all current track's data to drain before we say
- // that the track is stopped.
- if (mBytesRemaining == 0) {
- // Only start draining when all data in mixbuffer
- // has been written
- ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2");
- track->mState = TrackBase::STOPPING_2; // so presentation completes after drain
- // do not drain if no data was ever sent to HAL (mStandby == true)
- if (last && !mStandby) {
- // do not modify drain sequence if we are already draining. This happens
- // when resuming from pause after drain.
- if ((mDrainSequence & 1) == 0) {
- sleepTime = 0;
- standbyTime = systemTime() + standbyDelay;
- mixerStatus = MIXER_DRAIN_TRACK;
- mDrainSequence += 2;
- }
- if (mHwPaused) {
- // It is possible to move from PAUSED to STOPPING_1 without
- // a resume so we must ensure hardware is running
- doHwResume = true;
- mHwPaused = false;
- }
- }
- }
- } else if (track->isStopping_2()) {
- // Drain has completed or we are in standby, signal presentation complete
- if (!(mDrainSequence & 1) || !last || mStandby) {
- track->mState = TrackBase::STOPPED;
- size_t audioHALFrames =
- (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
- size_t framesWritten =
- mBytesWritten / audio_stream_out_frame_size(mOutput->stream);
- track->presentationComplete(framesWritten, audioHALFrames);
- track->reset();
- tracksToRemove->add(track);
- }
- } else {
- // No buffers for this track. Give it a few chances to
- // fill a buffer, then remove it from active list.
- if (--(track->mRetryCount) <= 0) {
- ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list",
- track->name());
- tracksToRemove->add(track);
- // indicate to client process that the track was disabled because of underrun;
- // it will then automatically call start() when data is available
- android_atomic_or(CBLK_DISABLED, &cblk->mFlags);
- } else if (last){
- mixerStatus = MIXER_TRACKS_ENABLED;
- }
- }
- }
- // compute volume for this track
- processVolume_l(track, last);
- }
- // make sure the pause/flush/resume sequence is executed in the right order.
- // If a flush is pending and a track is active but the HW is not paused, force a HW pause
- // before flush and then resume HW. This can happen in case of pause/flush/resume
- // if resume is received before pause is executed.
- if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
- mOutput->stream->pause(mOutput->stream);
- }
- if (mFlushPending) {
- flushHw_l();
- mFlushPending = false;
- }
- if (!mStandby && doHwResume) {
- mOutput->stream->resume(mOutput->stream);
- }
- // remove all the tracks that need to be...
- removeTracks_l(*tracksToRemove);
- return mixerStatus;
- }
- // must be called with thread mutex locked
- bool AudioFlinger::OffloadThread::waitingAsyncCallback_l()
- {
- ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d",
- mWriteAckSequence, mDrainSequence);
- if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) {
- return true;
- }
- return false;
- }
- // must be called with thread mutex locked
- bool AudioFlinger::OffloadThread::shouldStandby_l()
- {
- bool trackPaused = false;
- // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack
- // after a timeout and we will enter standby then.
- if (mTracks.size() > 0) {
- trackPaused = mTracks[mTracks.size() - 1]->isPaused();
- }
- return !mStandby && !trackPaused;
- }
- bool AudioFlinger::OffloadThread::waitingAsyncCallback()
- {
- Mutex::Autolock _l(mLock);
- return waitingAsyncCallback_l();
- }
- void AudioFlinger::OffloadThread::flushHw_l()
- {
- DirectOutputThread::flushHw_l();
- // Flush anything still waiting in the mixbuffer
- mCurrentWriteLength = 0;
- mBytesRemaining = 0;
- mPausedWriteLength = 0;
- mPausedBytesRemaining = 0;
- mHwPaused = false;
- if (mUseAsyncWrite) {
- // discard any pending drain or write ack by incrementing sequence
- mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
- mDrainSequence = (mDrainSequence + 2) & ~1;
- ALOG_ASSERT(mCallbackThread != 0);
- mCallbackThread->setWriteBlocked(mWriteAckSequence);
- mCallbackThread->setDraining(mDrainSequence);
- }
- }
- void AudioFlinger::OffloadThread::onAddNewTrack_l()
- {
- sp<Track> previousTrack = mPreviousTrack.promote();
- sp<Track> latestTrack = mLatestActiveTrack.promote();
- if (previousTrack != 0 && latestTrack != 0 &&
- (previousTrack->sessionId() != latestTrack->sessionId())) {
- mFlushPending = true;
- }
- PlaybackThread::onAddNewTrack_l();
- }
- void AudioFlinger::OffloadThread::onFatalError()
- {
- Mutex::Autolock _l(mLock);
- // call invalidate, to recreate track on fatal error
- invalidateTracks_l(AUDIO_STREAM_MUSIC);
- }
- // ----------------------------------------------------------------------------
- AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
- AudioFlinger::MixerThread* mainThread, audio_io_handle_t id)
- : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(),
- DUPLICATING),
- mWaitTimeMs(UINT_MAX)
- {
- addOutputTrack(mainThread);
- }
- AudioFlinger::DuplicatingThread::~DuplicatingThread()
- {
- for (size_t i = 0; i < mOutputTracks.size(); i++) {
- mOutputTracks[i]->destroy();
- }
- }
- void AudioFlinger::DuplicatingThread::threadLoop_mix()
- {
- // mix buffers...
- if (outputsReady(outputTracks)) {
- mAudioMixer->process(AudioBufferProvider::kInvalidPTS);
- } else {
- if (mMixerBufferValid) {
- memset(mMixerBuffer, 0, mMixerBufferSize);
- } else if (mEffectBufferValid) {
- memset(mEffectBuffer, 0, mEffectBufferSize);
- } else {
- memset(mSinkBuffer, 0, mSinkBufferSize);
- }
- }
- sleepTime = 0;
- writeFrames = mNormalFrameCount;
- mCurrentWriteLength = mSinkBufferSize;
- standbyTime = systemTime() + standbyDelay;
- }
- void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
- {
- if (sleepTime == 0) {
- if (mMixerStatus == MIXER_TRACKS_ENABLED) {
- sleepTime = activeSleepTime;
- } else {
- sleepTime = idleSleepTime;
- }
- } else if (mBytesWritten != 0) {
- if (mMixerStatus == MIXER_TRACKS_ENABLED) {
- writeFrames = mNormalFrameCount;
- if (mMixerBufferValid) {
- memset(mMixerBuffer, 0, mMixerBufferSize);
- } else {
- memset(mSinkBuffer, 0, mSinkBufferSize);
- }
- } else {
- // flush remaining overflow buffers in output tracks
- writeFrames = 0;
- }
- sleepTime = 0;
- }
- }
- ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
- {
- for (size_t i = 0; i < outputTracks.size(); i++) {
- // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT
- // for delivery downstream as needed. This in-place conversion is safe as
- // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format
- // (AUDIO_FORMAT_PCM_8_BIT is not allowed here).
- if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
- memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT,
- mSinkBuffer, mFormat, writeFrames * mChannelCount);
- }
- outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames);
- }
- mStandby = false;
- return (ssize_t)mSinkBufferSize;
- }
- void AudioFlinger::DuplicatingThread::threadLoop_standby()
- {
- // DuplicatingThread implements standby by stopping all tracks
- for (size_t i = 0; i < outputTracks.size(); i++) {
- outputTracks[i]->stop();
- }
- }
- void AudioFlinger::DuplicatingThread::saveOutputTracks()
- {
- outputTracks = mOutputTracks;
- }
- void AudioFlinger::DuplicatingThread::clearOutputTracks()
- {
- outputTracks.clear();
- }
- void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
- {
- Mutex::Autolock _l(mLock);
- // FIXME explain this formula
- size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
- // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat
- // due to current usage case and restrictions on the AudioBufferProvider.
- // Actual buffer conversion is done in threadLoop_write().
- //
- // TODO: This may change in the future, depending on multichannel
- // (and non int16_t*) support on AF::PlaybackThread::OutputTrack
- OutputTrack *outputTrack = new OutputTrack(thread,
- this,
- mSampleRate,
- AUDIO_FORMAT_PCM_16_BIT,
- mChannelMask,
- frameCount,
- IPCThreadState::self()->getCallingUid());
- if (outputTrack->cblk() != NULL) {
- thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
- mOutputTracks.add(outputTrack);
- ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
- updateWaitTime_l();
- }
- }
- void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
- {
- Mutex::Autolock _l(mLock);
- for (size_t i = 0; i < mOutputTracks.size(); i++) {
- if (mOutputTracks[i]->thread() == thread) {
- mOutputTracks[i]->destroy();
- mOutputTracks.removeAt(i);
- updateWaitTime_l();
- return;
- }
- }
- ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
- }
- // caller must hold mLock
- void AudioFlinger::DuplicatingThread::updateWaitTime_l()
- {
- mWaitTimeMs = UINT_MAX;
- for (size_t i = 0; i < mOutputTracks.size(); i++) {
- sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
- if (strong != 0) {
- uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
- if (waitTimeMs < mWaitTimeMs) {
- mWaitTimeMs = waitTimeMs;
- }
- }
- }
- }
- bool AudioFlinger::DuplicatingThread::outputsReady(
- const SortedVector< sp<OutputTrack> > &outputTracks)
- {
- for (size_t i = 0; i < outputTracks.size(); i++) {
- sp<ThreadBase> thread = outputTracks[i]->thread().promote();
- if (thread == 0) {
- ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p",
- outputTracks[i].get());
- return false;
- }
- PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
- // see note at standby() declaration
- if (playbackThread->standby() && !playbackThread->isSuspended()) {
- ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(),
- thread.get());
- return false;
- }
- }
- return true;
- }
- uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const
- {
- return (mWaitTimeMs * 1000) / 2;
- }
- void AudioFlinger::DuplicatingThread::cacheParameters_l()
- {
- // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first
- updateWaitTime_l();
- MixerThread::cacheParameters_l();
- }
- // ----------------------------------------------------------------------------
- // Record
- // ----------------------------------------------------------------------------
- AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
- AudioStreamIn *input,
- audio_io_handle_t id,
- audio_devices_t outDevice,
- audio_devices_t inDevice
- #ifdef TEE_SINK
- , const sp<NBAIO_Sink>& teeSink
- #endif
- ) :
- ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD),
- mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL),
- // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l()
- mRsmpInRear(0)
- #ifdef TEE_SINK
- , mTeeSink(teeSink)
- #endif
- , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize,
- "RecordThreadRO", MemoryHeapBase::READ_ONLY))
- // mFastCapture below
- , mFastCaptureFutex(0)
- // mInputSource
- // mPipeSink
- // mPipeSource
- , mPipeFramesP2(0)
- // mPipeMemory
- // mFastCaptureNBLogWriter
- , mFastTrackAvail(false)
- {
- snprintf(mName, kNameLength, "AudioIn_%X", id);
- mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName);
- readInputParameters_l();
- // create an NBAIO source for the HAL input stream, and negotiate
- mInputSource = new AudioStreamInSource(input->stream);
- size_t numCounterOffers = 0;
- const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)};
- ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers);
- ALOG_ASSERT(index == 0);
- // initialize fast capture depending on configuration
- bool initFastCapture;
- switch (kUseFastCapture) {
- case FastCapture_Never:
- initFastCapture = false;
- break;
- case FastCapture_Always:
- initFastCapture = true;
- break;
- case FastCapture_Static:
- uint32_t primaryOutputSampleRate;
- {
- AutoMutex _l(audioFlinger->mHardwareLock);
- primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate;
- }
- initFastCapture =
- // either capture sample rate is same as (a reasonable) primary output sample rate
- (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) &&
- (mSampleRate == primaryOutputSampleRate)) ||
- // or primary output sample rate is unknown, and capture sample rate is reasonable
- ((primaryOutputSampleRate == 0) &&
- ((mSampleRate == 44100 || mSampleRate == 48000)))) &&
- // and the buffer size is < 12 ms
- (mFrameCount * 1000) / mSampleRate < 12;
- break;
- // case FastCapture_Dynamic:
- }
- if (initFastCapture) {
- // create a Pipe for FastMixer to write to, and for us and fast tracks to read from
- NBAIO_Format format = mInputSource->format();
- size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each
- size_t pipeSize = pipeFramesP2 * Format_frameSize(format);
- void *pipeBuffer;
- const sp<MemoryDealer> roHeap(readOnlyHeap());
- sp<IMemory> pipeMemory;
- if ((roHeap == 0) ||
- (pipeMemory = roHeap->allocate(pipeSize)) == 0 ||
- (pipeBuffer = pipeMemory->pointer()) == NULL) {
- ALOGE("not enough memory for pipe buffer size=%zu", pipeSize);
- goto failed;
- }
- // pipe will be shared directly with fast clients, so clear to avoid leaking old information
- memset(pipeBuffer, 0, pipeSize);
- Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer);
- const NBAIO_Format offers[1] = {format};
- size_t numCounterOffers = 0;
- ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
- ALOG_ASSERT(index == 0);
- mPipeSink = pipe;
- PipeReader *pipeReader = new PipeReader(*pipe);
- numCounterOffers = 0;
- index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
- ALOG_ASSERT(index == 0);
- mPipeSource = pipeReader;
- mPipeFramesP2 = pipeFramesP2;
- mPipeMemory = pipeMemory;
- // create fast capture
- mFastCapture = new FastCapture();
- FastCaptureStateQueue *sq = mFastCapture->sq();
- #ifdef STATE_QUEUE_DUMP
- // FIXME
- #endif
- FastCaptureState *state = sq->begin();
- state->mCblk = NULL;
- state->mInputSource = mInputSource.get();
- state->mInputSourceGen++;
- state->mPipeSink = pipe;
- state->mPipeSinkGen++;
- state->mFrameCount = mFrameCount;
- state->mCommand = FastCaptureState::COLD_IDLE;
- // already done in constructor initialization list
- //mFastCaptureFutex = 0;
- state->mColdFutexAddr = &mFastCaptureFutex;
- state->mColdGen++;
- state->mDumpState = &mFastCaptureDumpState;
- #ifdef TEE_SINK
- // FIXME
- #endif
- mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture");
- state->mNBLogWriter = mFastCaptureNBLogWriter.get();
- sq->end();
- sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
- // start the fast capture
- mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO);
- pid_t tid = mFastCapture->getTid();
- int err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
- if (err != 0) {
- ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
- kPriorityFastCapture, getpid_cached, tid, err);
- }
- #ifdef AUDIO_WATCHDOG
- // FIXME
- #endif
- mFastTrackAvail = true;
- }
- failed: ;
- // FIXME mNormalSource
- }
- AudioFlinger::RecordThread::~RecordThread()
- {
- if (mFastCapture != 0) {
- FastCaptureStateQueue *sq = mFastCapture->sq();
- FastCaptureState *state = sq->begin();
- if (state->mCommand == FastCaptureState::COLD_IDLE) {
- int32_t old = android_atomic_inc(&mFastCaptureFutex);
- if (old == -1) {
- (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1);
- }
- }
- state->mCommand = FastCaptureState::EXIT;
- sq->end();
- sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
- mFastCapture->join();
- mFastCapture.clear();
- }
- mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter);
- mAudioFlinger->unregisterWriter(mNBLogWriter);
- delete[] mRsmpInBuffer;
- }
- void AudioFlinger::RecordThread::onFirstRef()
- {
- run(mName, PRIORITY_URGENT_AUDIO);
- }
- bool AudioFlinger::RecordThread::threadLoop()
- {
- nsecs_t lastWarning = 0;
- inputStandBy();
- reacquire_wakelock:
- sp<RecordTrack> activeTrack;
- int activeTracksGen;
- {
- Mutex::Autolock _l(mLock);
- size_t size = mActiveTracks.size();
- activeTracksGen = mActiveTracksGen;
- if (size > 0) {
- // FIXME an arbitrary choice
- activeTrack = mActiveTracks[0];
- acquireWakeLock_l(activeTrack->uid());
- if (size > 1) {
- SortedVector<int> tmp;
- for (size_t i = 0; i < size; i++) {
- tmp.add(mActiveTracks[i]->uid());
- }
- updateWakeLockUids_l(tmp);
- }
- } else {
- acquireWakeLock_l(-1);
- }
- }
- // used to request a deferred sleep, to be executed later while mutex is unlocked
- uint32_t sleepUs = 0;
- // loop while there is work to do
- for (;;) {
- Vector< sp<EffectChain> > effectChains;
- // sleep with mutex unlocked
- if (sleepUs > 0) {
- usleep(sleepUs);
- sleepUs = 0;
- }
- // activeTracks accumulates a copy of a subset of mActiveTracks
- Vector< sp<RecordTrack> > activeTracks;
- // reference to the (first and only) active fast track
- sp<RecordTrack> fastTrack;
- // reference to a fast track which is about to be removed
- sp<RecordTrack> fastTrackToRemove;
- { // scope for mLock
- Mutex::Autolock _l(mLock);
- processConfigEvents_l();
- // check exitPending here because checkForNewParameters_l() and
- // checkForNewParameters_l() can temporarily release mLock
- if (exitPending()) {
- break;
- }
- // if no active track(s), then standby and release wakelock
- size_t size = mActiveTracks.size();
- if (size == 0) {
- standbyIfNotAlreadyInStandby();
- // exitPending() can't become true here
- releaseWakeLock_l();
- ALOGV("RecordThread: loop stopping");
- // go to sleep
- mWaitWorkCV.wait(mLock);
- ALOGV("RecordThread: loop starting");
- goto reacquire_wakelock;
- }
- if (mActiveTracksGen != activeTracksGen) {
- activeTracksGen = mActiveTracksGen;
- SortedVector<int> tmp;
- for (size_t i = 0; i < size; i++) {
- tmp.add(mActiveTracks[i]->uid());
- }
- updateWakeLockUids_l(tmp);
- }
- bool doBroadcast = false;
- for (size_t i = 0; i < size; ) {
- activeTrack = mActiveTracks[i];
- if (activeTrack->isTerminated()) {
- if (activeTrack->isFastTrack()) {
- ALOG_ASSERT(fastTrackToRemove == 0);
- fastTrackToRemove = activeTrack;
- }
- removeTrack_l(activeTrack);
- mActiveTracks.remove(activeTrack);
- mActiveTracksGen++;
- size--;
- continue;
- }
- TrackBase::track_state activeTrackState = activeTrack->mState;
- switch (activeTrackState) {
- case TrackBase::PAUSING:
- mActiveTracks.remove(activeTrack);
- mActiveTracksGen++;
- doBroadcast = true;
- size--;
- continue;
- case TrackBase::STARTING_1:
- sleepUs = 10000;
- i++;
- continue;
- case TrackBase::STARTING_2:
- doBroadcast = true;
- mStandby = false;
- activeTrack->mState = TrackBase::ACTIVE;
- break;
- case TrackBase::ACTIVE:
- break;
- case TrackBase::IDLE:
- i++;
- continue;
- default:
- LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState);
- }
- activeTracks.add(activeTrack);
- i++;
- if (activeTrack->isFastTrack()) {
- ALOG_ASSERT(!mFastTrackAvail);
- ALOG_ASSERT(fastTrack == 0);
- fastTrack = activeTrack;
- }
- }
- if (doBroadcast) {
- mStartStopCond.broadcast();
- }
- // sleep if there are no active tracks to process
- if (activeTracks.size() == 0) {
- if (sleepUs == 0) {
- sleepUs = kRecordThreadSleepUs;
- }
- continue;
- }
- sleepUs = 0;
- lockEffectChains_l(effectChains);
- }
- // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0
- size_t size = effectChains.size();
- for (size_t i = 0; i < size; i++) {
- // thread mutex is not locked, but effect chain is locked
- effectChains[i]->process_l();
- }
- // Push a new fast capture state if fast capture is not already running, or cblk change
- if (mFastCapture != 0) {
- FastCaptureStateQueue *sq = mFastCapture->sq();
- FastCaptureState *state = sq->begin();
- bool didModify = false;
- FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED;
- if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME &&
- (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) {
- if (state->mCommand == FastCaptureState::COLD_IDLE) {
- int32_t old = android_atomic_inc(&mFastCaptureFutex);
- if (old == -1) {
- (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1);
- }
- }
- state->mCommand = FastCaptureState::READ_WRITE;
- #if 0 // FIXME
- mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
- FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN);
- #endif
- didModify = true;
- }
- audio_track_cblk_t *cblkOld = state->mCblk;
- audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL;
- if (cblkNew != cblkOld) {
- state->mCblk = cblkNew;
- // block until acked if removing a fast track
- if (cblkOld != NULL) {
- block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED;
- }
- didModify = true;
- }
- sq->end(didModify);
- if (didModify) {
- sq->push(block);
- #if 0
- if (kUseFastCapture == FastCapture_Dynamic) {
- mNormalSource = mPipeSource;
- }
- #endif
- }
- }
- // now run the fast track destructor with thread mutex unlocked
- fastTrackToRemove.clear();
- // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one.
- // Only the client(s) that are too slow will overrun. But if even the fastest client is too
- // slow, then this RecordThread will overrun by not calling HAL read often enough.
- // If destination is non-contiguous, first read past the nominal end of buffer, then
- // copy to the right place. Permitted because mRsmpInBuffer was over-allocated.
- // Use modulo operator instead of and operator.
- // x &= (y-1) returns the remainder if y is even
- // Use modulo operator to generalize it for all values.
- // This is needed for compress offload voip and encode usecases.
- int32_t rear = mRsmpInRear % mRsmpInFramesP2;
- ssize_t framesRead;
- // If an NBAIO source is present, use it to read the normal capture's data
- if (mPipeSource != 0) {
- size_t framesToRead = mBufferSize / mFrameSize;
- framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount],
- framesToRead, AudioBufferProvider::kInvalidPTS);
- if (framesRead == 0) {
- // since pipe is non-blocking, simulate blocking input
- sleepUs = (framesToRead * 1000000LL) / mSampleRate;
- }
- // otherwise use the HAL / AudioStreamIn directly
- } else {
- ssize_t bytesRead = mInput->stream->read(mInput->stream,
- &mRsmpInBuffer[rear * mChannelCount], mBufferSize);
- if (bytesRead < 0) {
- framesRead = bytesRead;
- } else {
- framesRead = bytesRead / mFrameSize;
- }
- }
- if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) {
- ALOGE("read failed: framesRead=%d", framesRead);
- // Force input into standby so that it tries to recover at next read attempt
- inputStandBy();
- sleepUs = kRecordThreadSleepUs;
- }
- if (framesRead <= 0) {
- goto unlock;
- }
- ALOG_ASSERT(framesRead > 0);
- if (mTeeSink != 0) {
- (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead);
- }
- // If destination is non-contiguous, we now correct for reading past end of buffer.
- {
- size_t part1 = mRsmpInFramesP2 - rear;
- if ((size_t) framesRead > part1) {
- memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount],
- (framesRead - part1) * mFrameSize);
- }
- }
- rear = mRsmpInRear += framesRead;
- size = activeTracks.size();
- // loop over each active track
- for (size_t i = 0; i < size; i++) {
- activeTrack = activeTracks[i];
- // skip fast tracks, as those are handled directly by FastCapture
- if (activeTrack->isFastTrack()) {
- continue;
- }
- enum {
- OVERRUN_UNKNOWN,
- OVERRUN_TRUE,
- OVERRUN_FALSE
- } overrun = OVERRUN_UNKNOWN;
- // loop over getNextBuffer to handle circular sink
- for (;;) {
- activeTrack->mSink.frameCount = ~0;
- status_t status = activeTrack->getNextBuffer(&activeTrack->mSink);
- size_t framesOut = activeTrack->mSink.frameCount;
- LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0));
- int32_t front = activeTrack->mRsmpInFront;
- ssize_t filled = rear - front;
- size_t framesIn;
- if (filled < 0) {
- // should not happen, but treat like a massive overrun and re-sync
- framesIn = 0;
- activeTrack->mRsmpInFront = rear;
- overrun = OVERRUN_TRUE;
- } else if ((size_t) filled <= mRsmpInFrames) {
- framesIn = (size_t) filled;
- } else {
- // client is not keeping up with server, but give it latest data
- framesIn = mRsmpInFrames;
- activeTrack->mRsmpInFront = front = rear - framesIn;
- overrun = OVERRUN_TRUE;
- }
- if (framesOut == 0 || framesIn == 0) {
- break;
- }
- if (activeTrack->mResampler == NULL) {
- // no resampling
- if (framesIn > framesOut) {
- framesIn = framesOut;
- } else {
- framesOut = framesIn;
- }
- int8_t *dst = activeTrack->mSink.i8;
- while (framesIn > 0) {
- // Use modulo operator instead of and operator.
- // x &= (y-1) returns the remainder if y is even
- // Use modulo operator to generalize it for all values.
- // This is needed for compress offload voip and encode usecases.
- front %= mRsmpInFramesP2;
- size_t part1 = mRsmpInFramesP2 - front;
- if (part1 > framesIn) {
- part1 = framesIn;
- }
- int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize);
- if (mChannelCount == activeTrack->mChannelCount ||
- audio_is_compress_capture_format(mFormat) ||
- audio_is_compress_voip_format(mFormat)) {
- memcpy(dst, src, part1 * mFrameSize);
- } else if (mChannelCount == 1) {
- upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src,
- part1);
- } else {
- downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src,
- part1);
- }
- dst += part1 * activeTrack->mFrameSize;
- front += part1;
- framesIn -= part1;
- }
- activeTrack->mRsmpInFront += framesOut;
- } else {
- // resampling
- // FIXME framesInNeeded should really be part of resampler API, and should
- // depend on the SRC ratio
- // to keep mRsmpInBuffer full so resampler always has sufficient input
- size_t framesInNeeded;
- // FIXME only re-calculate when it changes, and optimize for common ratios
- // Do not precompute in/out because floating point is not associative
- // e.g. a*b/c != a*(b/c).
- const double in(mSampleRate);
- const double out(activeTrack->mSampleRate);
- framesInNeeded = ceil(framesOut * in / out) + 1;
- ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g",
- framesInNeeded, framesOut, in / out);
- // Although we theoretically have framesIn in circular buffer, some of those are
- // unreleased frames, and thus must be discounted for purpose of budgeting.
- size_t unreleased = activeTrack->mRsmpInUnrel;
- framesIn = framesIn > unreleased ? framesIn - unreleased : 0;
- if (framesIn < framesInNeeded) {
- ALOGV("not enough to resample: have %u frames in but need %u in to "
- "produce %u out given in/out ratio of %.4g",
- framesIn, framesInNeeded, framesOut, in / out);
- size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0;
- LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut);
- if (newFramesOut == 0) {
- break;
- }
- framesInNeeded = ceil(newFramesOut * in / out) + 1;
- ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g",
- framesInNeeded, newFramesOut, out / in);
- LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded);
- ALOGV("success 2: have %u frames in and need %u in to produce %u out "
- "given in/out ratio of %.4g",
- framesIn, framesInNeeded, newFramesOut, in / out);
- framesOut = newFramesOut;
- } else {
- ALOGV("success 1: have %u in and need %u in to produce %u out "
- "given in/out ratio of %.4g",
- framesIn, framesInNeeded, framesOut, in / out);
- }
- // reallocate mRsmpOutBuffer as needed; we will grow but never shrink
- if (activeTrack->mRsmpOutFrameCount < framesOut) {
- // FIXME why does each track need it's own mRsmpOutBuffer? can't they share?
- delete[] activeTrack->mRsmpOutBuffer;
- // resampler always outputs stereo
- activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2];
- activeTrack->mRsmpOutFrameCount = framesOut;
- }
- // resampler accumulates, but we only have one source track
- memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t));
- activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut,
- // FIXME how about having activeTrack implement this interface itself?
- activeTrack->mResamplerBufferProvider
- /*this*/ /* AudioBufferProvider* */);
- // ditherAndClamp() works as long as all buffers returned by
- // activeTrack->getNextBuffer() are 32 bit aligned which should be always true.
- if (activeTrack->mChannelCount == 1) {
- // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t
- ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer,
- framesOut);
- // the resampler always outputs stereo samples:
- // do post stereo to mono conversion
- downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16,
- (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut);
- } else {
- ditherAndClamp((int32_t *)activeTrack->mSink.raw,
- activeTrack->mRsmpOutBuffer, framesOut);
- }
- // now done with mRsmpOutBuffer
- }
- if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) {
- overrun = OVERRUN_FALSE;
- }
- if (activeTrack->mFramesToDrop == 0) {
- if (framesOut > 0) {
- activeTrack->mSink.frameCount = framesOut;
- activeTrack->releaseBuffer(&activeTrack->mSink);
- }
- } else {
- // FIXME could do a partial drop of framesOut
- if (activeTrack->mFramesToDrop > 0) {
- activeTrack->mFramesToDrop -= framesOut;
- if (activeTrack->mFramesToDrop <= 0) {
- activeTrack->clearSyncStartEvent();
- }
- } else {
- activeTrack->mFramesToDrop += framesOut;
- if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 ||
- activeTrack->mSyncStartEvent->isCancelled()) {
- ALOGW("Synced record %s, session %d, trigger session %d",
- (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled",
- activeTrack->sessionId(),
- (activeTrack->mSyncStartEvent != 0) ?
- activeTrack->mSyncStartEvent->triggerSession() : 0);
- activeTrack->clearSyncStartEvent();
- }
- }
- }
- if (framesOut == 0) {
- break;
- }
- }
- switch (overrun) {
- case OVERRUN_TRUE:
- // client isn't retrieving buffers fast enough
- if (!activeTrack->setOverflow()) {
- nsecs_t now = systemTime();
- // FIXME should lastWarning per track?
- if ((now - lastWarning) > kWarningThrottleNs) {
- ALOGW("RecordThread: buffer overflow");
- lastWarning = now;
- }
- }
- break;
- case OVERRUN_FALSE:
- activeTrack->clearOverflow();
- break;
- case OVERRUN_UNKNOWN:
- break;
- }
- }
- unlock:
- // enable changes in effect chain
- unlockEffectChains(effectChains);
- // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end
- }
- standbyIfNotAlreadyInStandby();
- {
- Mutex::Autolock _l(mLock);
- for (size_t i = 0; i < mTracks.size(); i++) {
- sp<RecordTrack> track = mTracks[i];
- track->invalidate();
- }
- mActiveTracks.clear();
- mActiveTracksGen++;
- mStartStopCond.broadcast();
- }
- releaseWakeLock();
- ALOGV("RecordThread %p exiting", this);
- return false;
- }
- void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby()
- {
- if (!mStandby) {
- inputStandBy();
- mStandby = true;
- }
- }
- void AudioFlinger::RecordThread::inputStandBy()
- {
- // Idle the fast capture if it's currently running
- if (mFastCapture != 0) {
- FastCaptureStateQueue *sq = mFastCapture->sq();
- FastCaptureState *state = sq->begin();
- if (!(state->mCommand & FastCaptureState::IDLE)) {
- state->mCommand = FastCaptureState::COLD_IDLE;
- state->mColdFutexAddr = &mFastCaptureFutex;
- state->mColdGen++;
- mFastCaptureFutex = 0;
- sq->end();
- // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
- sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED);
- #if 0
- if (kUseFastCapture == FastCapture_Dynamic) {
- // FIXME
- }
- #endif
- #ifdef AUDIO_WATCHDOG
- // FIXME
- #endif
- } else {
- sq->end(false /*didModify*/);
- }
- }
- mInput->stream->common.standby(&mInput->stream->common);
- }
- // RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held
- sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
- const sp<AudioFlinger::Client>& client,
- uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- size_t *pFrameCount,
- int sessionId,
- size_t *notificationFrames,
- int uid,
- IAudioFlinger::track_flags_t *flags,
- pid_t tid,
- status_t *status)
- {
- size_t frameCount = *pFrameCount;
- sp<RecordTrack> track;
- status_t lStatus;
- // client expresses a preference for FAST, but we get the final say
- if (*flags & IAudioFlinger::TRACK_FAST) {
- if (
- // use case: callback handler
- (tid != -1) &&
- // frame count is not specified, or is exactly the pipe depth
- ((frameCount == 0) || (frameCount == mPipeFramesP2)) &&
- // PCM data
- audio_is_linear_pcm(format) &&
- // native format
- (format == mFormat) &&
- // native channel mask
- (channelMask == mChannelMask) &&
- // native hardware sample rate
- (sampleRate == mSampleRate) &&
- // record thread has an associated fast capture
- hasFastCapture() &&
- // there are sufficient fast track slots available
- mFastTrackAvail
- ) {
- ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u",
- frameCount, mFrameCount);
- } else {
- ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u "
- "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
- "hasFastCapture=%d tid=%d mFastTrackAvail=%d",
- frameCount, mFrameCount, mPipeFramesP2,
- format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate,
- hasFastCapture(), tid, mFastTrackAvail);
- *flags &= ~IAudioFlinger::TRACK_FAST;
- }
- }
- // compute track buffer size in frames, and suggest the notification frame count
- if (*flags & IAudioFlinger::TRACK_FAST) {
- // fast track: frame count is exactly the pipe depth
- frameCount = mPipeFramesP2;
- // ignore requested notificationFrames, and always notify exactly once every HAL buffer
- *notificationFrames = mFrameCount;
- } else {
- // not fast track: max notification period is resampled equivalent of one HAL buffer time
- // or 20 ms if there is a fast capture
- // TODO This could be a roundupRatio inline, and const
- size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount)
- * sampleRate + mSampleRate - 1) / mSampleRate;
- // minimum number of notification periods is at least kMinNotifications,
- // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs)
- static const size_t kMinNotifications = 3;
- static const uint32_t kMinMs = 30;
- // TODO This could be a roundupRatio inline
- const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000;
- // TODO This could be a roundupRatio inline
- const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) /
- maxNotificationFrames;
- const size_t minFrameCount = maxNotificationFrames *
- max(kMinNotifications, minNotificationsByMs);
- frameCount = max(frameCount, minFrameCount);
- if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) {
- *notificationFrames = maxNotificationFrames;
- }
- }
- *pFrameCount = frameCount;
- lStatus = initCheck();
- if (lStatus != NO_ERROR) {
- ALOGE("createRecordTrack_l() audio driver not initialized");
- goto Exit;
- }
- { // scope for mLock
- Mutex::Autolock _l(mLock);
- track = new RecordTrack(this, client, sampleRate,
- format, channelMask, frameCount, NULL, sessionId, uid,
- *flags, TrackBase::TYPE_DEFAULT);
- lStatus = track->initCheck();
- if (lStatus != NO_ERROR) {
- ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus);
- // track must be cleared from the caller as the caller has the AF lock
- goto Exit;
- }
- mTracks.add(track);
- // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
- bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
- mAudioFlinger->btNrecIsOff();
- setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
- setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
- if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) {
- pid_t callingPid = IPCThreadState::self()->getCallingPid();
- // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
- // so ask activity manager to do this on our behalf
- sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp);
- }
- }
- lStatus = NO_ERROR;
- Exit:
- *status = lStatus;
- return track;
- }
- status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack,
- AudioSystem::sync_event_t event,
- int triggerSession)
- {
- ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession);
- sp<ThreadBase> strongMe = this;
- status_t status = NO_ERROR;
- if (event == AudioSystem::SYNC_EVENT_NONE) {
- recordTrack->clearSyncStartEvent();
- } else if (event != AudioSystem::SYNC_EVENT_SAME) {
- recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event,
- triggerSession,
- recordTrack->sessionId(),
- syncStartEventCallback,
- recordTrack);
- // Sync event can be cancelled by the trigger session if the track is not in a
- // compatible state in which case we start record immediately
- if (recordTrack->mSyncStartEvent->isCancelled()) {
- recordTrack->clearSyncStartEvent();
- } else {
- // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs
- recordTrack->mFramesToDrop = -
- ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000);
- }
- }
- {
- // This section is a rendezvous between binder thread executing start() and RecordThread
- AutoMutex lock(mLock);
- if (mActiveTracks.indexOf(recordTrack) >= 0) {
- if (recordTrack->mState == TrackBase::PAUSING) {
- ALOGV("active record track PAUSING -> ACTIVE");
- recordTrack->mState = TrackBase::ACTIVE;
- } else {
- ALOGV("active record track state %d", recordTrack->mState);
- }
- return status;
- }
- // TODO consider other ways of handling this, such as changing the state to :STARTING and
- // adding the track to mActiveTracks after returning from AudioSystem::startInput(),
- // or using a separate command thread
- recordTrack->mState = TrackBase::STARTING_1;
- mActiveTracks.add(recordTrack);
- mActiveTracksGen++;
- status_t status = NO_ERROR;
- if (recordTrack->isExternalTrack()) {
- mLock.unlock();
- status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId());
- mLock.lock();
- // FIXME should verify that recordTrack is still in mActiveTracks
- if (status != NO_ERROR) {
- mActiveTracks.remove(recordTrack);
- mActiveTracksGen++;
- recordTrack->clearSyncStartEvent();
- ALOGV("RecordThread::start error %d", status);
- return status;
- }
- }
- // Catch up with current buffer indices if thread is already running.
- // This is what makes a new client discard all buffered data. If the track's mRsmpInFront
- // was initialized to some value closer to the thread's mRsmpInFront, then the track could
- // see previously buffered data before it called start(), but with greater risk of overrun.
- recordTrack->mRsmpInFront = mRsmpInRear;
- recordTrack->mRsmpInUnrel = 0;
- // FIXME why reset?
- if (recordTrack->mResampler != NULL) {
- recordTrack->mResampler->reset();
- }
- recordTrack->mState = TrackBase::STARTING_2;
- // signal thread to start
- mWaitWorkCV.broadcast();
- if (mActiveTracks.indexOf(recordTrack) < 0) {
- ALOGV("Record failed to start");
- status = BAD_VALUE;
- goto startError;
- }
- return status;
- }
- startError:
- if (recordTrack->isExternalTrack()) {
- AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId());
- }
- recordTrack->clearSyncStartEvent();
- // FIXME I wonder why we do not reset the state here?
- return status;
- }
- void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event)
- {
- sp<SyncEvent> strongEvent = event.promote();
- if (strongEvent != 0) {
- sp<RefBase> ptr = strongEvent->cookie().promote();
- if (ptr != 0) {
- RecordTrack *recordTrack = (RecordTrack *)ptr.get();
- recordTrack->handleSyncStartEvent(strongEvent);
- }
- }
- }
- bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
- ALOGV("RecordThread::stop");
- AutoMutex _l(mLock);
- if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) {
- return false;
- }
- // note that threadLoop may still be processing the track at this point [without lock]
- recordTrack->mState = TrackBase::PAUSING;
- // do not wait for mStartStopCond if exiting
- if (exitPending()) {
- return true;
- }
- // FIXME incorrect usage of wait: no explicit predicate or loop
- mStartStopCond.wait(mLock);
- // if we have been restarted, recordTrack is in mActiveTracks here
- if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) {
- ALOGV("Record stopped OK");
- return true;
- }
- return false;
- }
- bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const
- {
- return false;
- }
- status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused)
- {
- #if 0 // This branch is currently dead code, but is preserved in case it will be needed in future
- if (!isValidSyncEvent(event)) {
- return BAD_VALUE;
- }
- int eventSession = event->triggerSession();
- status_t ret = NAME_NOT_FOUND;
- Mutex::Autolock _l(mLock);
- for (size_t i = 0; i < mTracks.size(); i++) {
- sp<RecordTrack> track = mTracks[i];
- if (eventSession == track->sessionId()) {
- (void) track->setSyncEvent(event);
- ret = NO_ERROR;
- }
- }
- return ret;
- #else
- return BAD_VALUE;
- #endif
- }
- // destroyTrack_l() must be called with ThreadBase::mLock held
- void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track)
- {
- track->terminate();
- track->mState = TrackBase::STOPPED;
- // active tracks are removed by threadLoop()
- if (mActiveTracks.indexOf(track) < 0) {
- removeTrack_l(track);
- }
- }
- void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track)
- {
- mTracks.remove(track);
- // need anything related to effects here?
- if (track->isFastTrack()) {
- ALOG_ASSERT(!mFastTrackAvail);
- mFastTrackAvail = true;
- }
- }
- void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
- {
- dumpInternals(fd, args);
- dumpTracks(fd, args);
- dumpEffectChains(fd, args);
- }
- void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args)
- {
- dprintf(fd, "\nInput thread %p:\n", this);
- if (mActiveTracks.size() > 0) {
- dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize);
- } else {
- dprintf(fd, " No active record clients\n");
- }
- dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no");
- dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no");
- dumpBase(fd, args);
- }
- void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused)
- {
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
- size_t numtracks = mTracks.size();
- size_t numactive = mActiveTracks.size();
- size_t numactiveseen = 0;
- dprintf(fd, " %d Tracks", numtracks);
- if (numtracks) {
- dprintf(fd, " of which %d are active\n", numactive);
- RecordTrack::appendDumpHeader(result);
- for (size_t i = 0; i < numtracks ; ++i) {
- sp<RecordTrack> track = mTracks[i];
- if (track != 0) {
- bool active = mActiveTracks.indexOf(track) >= 0;
- if (active) {
- numactiveseen++;
- }
- track->dump(buffer, SIZE, active);
- result.append(buffer);
- }
- }
- } else {
- dprintf(fd, "\n");
- }
- if (numactiveseen != numactive) {
- snprintf(buffer, SIZE, " The following tracks are in the active list but"
- " not in the track list\n");
- result.append(buffer);
- RecordTrack::appendDumpHeader(result);
- for (size_t i = 0; i < numactive; ++i) {
- sp<RecordTrack> track = mActiveTracks[i];
- if (mTracks.indexOf(track) < 0) {
- track->dump(buffer, SIZE, true);
- result.append(buffer);
- }
- }
- }
- write(fd, result.string(), result.size());
- }
- // AudioBufferProvider interface
- status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
- AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
- {
- RecordTrack *activeTrack = mRecordTrack;
- sp<ThreadBase> threadBase = activeTrack->mThread.promote();
- if (threadBase == 0) {
- buffer->frameCount = 0;
- buffer->raw = NULL;
- return NOT_ENOUGH_DATA;
- }
- RecordThread *recordThread = (RecordThread *) threadBase.get();
- int32_t rear = recordThread->mRsmpInRear;
- int32_t front = activeTrack->mRsmpInFront;
- ssize_t filled = rear - front;
- // FIXME should not be P2 (don't want to increase latency)
- // FIXME if client not keeping up, discard
- LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames));
- // 'filled' may be non-contiguous, so return only the first contiguous chunk
- front &= recordThread->mRsmpInFramesP2 - 1;
- size_t part1 = recordThread->mRsmpInFramesP2 - front;
- if (part1 > (size_t) filled) {
- part1 = filled;
- }
- size_t ask = buffer->frameCount;
- ALOG_ASSERT(ask > 0);
- if (part1 > ask) {
- part1 = ask;
- }
- if (part1 == 0) {
- // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty
- LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved");
- buffer->raw = NULL;
- buffer->frameCount = 0;
- activeTrack->mRsmpInUnrel = 0;
- return NOT_ENOUGH_DATA;
- }
- buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount;
- buffer->frameCount = part1;
- activeTrack->mRsmpInUnrel = part1;
- return NO_ERROR;
- }
- // AudioBufferProvider interface
- void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer(
- AudioBufferProvider::Buffer* buffer)
- {
- RecordTrack *activeTrack = mRecordTrack;
- size_t stepCount = buffer->frameCount;
- if (stepCount == 0) {
- return;
- }
- ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel);
- activeTrack->mRsmpInUnrel -= stepCount;
- activeTrack->mRsmpInFront += stepCount;
- buffer->raw = NULL;
- buffer->frameCount = 0;
- }
- bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair,
- status_t& status)
- {
- bool reconfig = false;
- status = NO_ERROR;
- audio_format_t reqFormat = mFormat;
- uint32_t samplingRate = mSampleRate;
- audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount);
- AudioParameter param = AudioParameter(keyValuePair);
- int value;
- // TODO Investigate when this code runs. Check with audio policy when a sample rate and
- // channel count change can be requested. Do we mandate the first client defines the
- // HAL sampling rate and channel count or do we allow changes on the fly?
- if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
- samplingRate = value;
- reconfig = true;
- }
- if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
- if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
- status = BAD_VALUE;
- } else {
- reqFormat = (audio_format_t) value;
- reconfig = true;
- }
- }
- if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
- audio_channel_mask_t mask = (audio_channel_mask_t) value;
- if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) {
- status = BAD_VALUE;
- } else {
- channelMask = mask;
- reconfig = true;
- }
- }
- if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
- // do not accept frame count changes if tracks are open as the track buffer
- // size depends on frame count and correct behavior would not be guaranteed
- // if frame count is changed after track creation
- if (mActiveTracks.size() > 0) {
- status = INVALID_OPERATION;
- } else {
- reconfig = true;
- }
- }
- if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
- // forward device change to effects that have requested to be
- // aware of attached audio device.
- for (size_t i = 0; i < mEffectChains.size(); i++) {
- mEffectChains[i]->setDevice_l(value);
- }
- // store input device and output device but do not forward output device to audio HAL.
- // Note that status is ignored by the caller for output device
- // (see AudioFlinger::setParameters()
- if (audio_is_output_devices(value)) {
- mOutDevice = value;
- status = BAD_VALUE;
- } else {
- mInDevice = value;
- // disable AEC and NS if the device is a BT SCO headset supporting those
- // pre processings
- if (mTracks.size() > 0) {
- bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
- mAudioFlinger->btNrecIsOff();
- for (size_t i = 0; i < mTracks.size(); i++) {
- sp<RecordTrack> track = mTracks[i];
- setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId());
- setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId());
- }
- }
- }
- }
- if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR &&
- mAudioSource != (audio_source_t)value) {
- // forward device change to effects that have requested to be
- // aware of attached audio device.
- for (size_t i = 0; i < mEffectChains.size(); i++) {
- mEffectChains[i]->setAudioSource_l((audio_source_t)value);
- }
- mAudioSource = (audio_source_t)value;
- }
- if (status == NO_ERROR) {
- status = mInput->stream->common.set_parameters(&mInput->stream->common,
- keyValuePair.string());
- if (status == INVALID_OPERATION) {
- inputStandBy();
- status = mInput->stream->common.set_parameters(&mInput->stream->common,
- keyValuePair.string());
- }
- if (reconfig) {
- if (status == BAD_VALUE &&
- reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
- reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
- (mInput->stream->common.get_sample_rate(&mInput->stream->common)
- <= (2 * samplingRate)) &&
- audio_channel_count_from_in_mask(
- mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 &&
- (channelMask == AUDIO_CHANNEL_IN_MONO ||
- channelMask == AUDIO_CHANNEL_IN_STEREO)) {
- status = NO_ERROR;
- }
- if (status == NO_ERROR) {
- readInputParameters_l();
- sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
- }
- }
- }
- return reconfig;
- }
- String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
- {
- Mutex::Autolock _l(mLock);
- if (initCheck() != NO_ERROR) {
- return String8();
- }
- char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
- const String8 out_s8(s);
- free(s);
- return out_s8;
- }
- void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) {
- AudioSystem::OutputDescriptor desc;
- const void *param2 = NULL;
- switch (event) {
- case AudioSystem::INPUT_OPENED:
- case AudioSystem::INPUT_CONFIG_CHANGED:
- desc.channelMask = mChannelMask;
- desc.samplingRate = mSampleRate;
- desc.format = mFormat;
- desc.frameCount = mFrameCount;
- desc.latency = 0;
- param2 = &desc;
- break;
- case AudioSystem::INPUT_CLOSED:
- default:
- break;
- }
- mAudioFlinger->audioConfigChanged(event, mId, param2);
- }
- void AudioFlinger::RecordThread::readInputParameters_l()
- {
- mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
- mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
- mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
- mHALFormat = mInput->stream->common.get_format(&mInput->stream->common);
- mFormat = mHALFormat;
- if (mFormat != AUDIO_FORMAT_PCM_16_BIT &&
- !audio_is_compress_voip_format(mFormat) &&
- !audio_is_compress_capture_format(mFormat)) {
- ALOGE("HAL format %#x not supported;", mFormat);
- }
- mFrameSize = audio_stream_in_frame_size(mInput->stream);
- mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common);
- mFrameCount = mBufferSize / mFrameSize;
- // This is the formula for calculating the temporary buffer size.
- // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to
- // 1 full output buffer, regardless of the alignment of the available input.
- // The value is somewhat arbitrary, and could probably be even larger.
- // A larger value should allow more old data to be read after a track calls start(),
- // without increasing latency.
- if (audio_is_compress_voip_format(mFormat) ||
- audio_is_compress_capture_format(mFormat)) {
- mRsmpInFrames = mFrameCount;
- mRsmpInFramesP2 = mRsmpInFrames;
- } else {
- mRsmpInFrames = mFrameCount * 7;
- mRsmpInFramesP2 = roundup(mRsmpInFrames);
- }
- delete[] mRsmpInBuffer;
- // TODO optimize audio capture buffer sizes ...
- // Here we calculate the size of the sliding buffer used as a source
- // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7).
- // For current HAL frame counts, this is usually 2048 = 40 ms. It would
- // be better to have it derived from the pipe depth in the long term.
- // The current value is higher than necessary. However it should not add to latency.
- // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer
- mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount];
- // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints.
- // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks?
- }
- uint32_t AudioFlinger::RecordThread::getInputFramesLost()
- {
- Mutex::Autolock _l(mLock);
- if (initCheck() != NO_ERROR) {
- return 0;
- }
- return mInput->stream->get_input_frames_lost(mInput->stream);
- }
- uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const
- {
- Mutex::Autolock _l(mLock);
- uint32_t result = 0;
- if (getEffectChain_l(sessionId) != 0) {
- result = EFFECT_SESSION;
- }
- for (size_t i = 0; i < mTracks.size(); ++i) {
- if (sessionId == mTracks[i]->sessionId()) {
- result |= TRACK_SESSION;
- break;
- }
- }
- return result;
- }
- KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const
- {
- KeyedVector<int, bool> ids;
- Mutex::Autolock _l(mLock);
- for (size_t j = 0; j < mTracks.size(); ++j) {
- sp<RecordThread::RecordTrack> track = mTracks[j];
- int sessionId = track->sessionId();
- if (ids.indexOfKey(sessionId) < 0) {
- ids.add(sessionId, true);
- }
- }
- return ids;
- }
- AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
- {
- Mutex::Autolock _l(mLock);
- AudioStreamIn *input = mInput;
- mInput = NULL;
- return input;
- }
- // this method must always be called either with ThreadBase mLock held or inside the thread loop
- audio_stream_t* AudioFlinger::RecordThread::stream() const
- {
- if (mInput == NULL) {
- return NULL;
- }
- return &mInput->stream->common;
- }
- status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
- {
- // only one chain per input thread
- if (mEffectChains.size() != 0) {
- ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this);
- return INVALID_OPERATION;
- }
- ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
- chain->setThread(this);
- chain->setInBuffer(NULL);
- chain->setOutBuffer(NULL);
- checkSuspendOnAddEffectChain_l(chain);
- // make sure enabled pre processing effects state is communicated to the HAL as we
- // just moved them to a new input stream.
- chain->syncHalEffectsState();
- mEffectChains.add(chain);
- return NO_ERROR;
- }
- size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
- {
- ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
- ALOGW_IF(mEffectChains.size() != 1,
- "removeEffectChain_l() %p invalid chain size %d on thread %p",
- chain.get(), mEffectChains.size(), this);
- if (mEffectChains.size() == 1) {
- mEffectChains.removeAt(0);
- }
- return 0;
- }
- status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch,
- audio_patch_handle_t *handle)
- {
- status_t status = NO_ERROR;
- if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
- // store new device and send to effects
- mInDevice = patch->sources[0].ext.device.type;
- for (size_t i = 0; i < mEffectChains.size(); i++) {
- mEffectChains[i]->setDevice_l(mInDevice);
- }
- // disable AEC and NS if the device is a BT SCO headset supporting those
- // pre processings
- if (mTracks.size() > 0) {
- bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
- mAudioFlinger->btNrecIsOff();
- for (size_t i = 0; i < mTracks.size(); i++) {
- sp<RecordTrack> track = mTracks[i];
- setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId());
- setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId());
- }
- }
- // store new source and send to effects
- if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) {
- mAudioSource = patch->sinks[0].ext.mix.usecase.source;
- for (size_t i = 0; i < mEffectChains.size(); i++) {
- mEffectChains[i]->setAudioSource_l(mAudioSource);
- }
- }
- audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice();
- status = hwDevice->create_audio_patch(hwDevice,
- patch->num_sources,
- patch->sources,
- patch->num_sinks,
- patch->sinks,
- handle);
- } else {
- ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL");
- }
- return status;
- }
- status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
- {
- status_t status = NO_ERROR;
- if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
- audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice();
- status = hwDevice->release_audio_patch(hwDevice, handle);
- } else {
- ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL");
- }
- return status;
- }
- void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record)
- {
- Mutex::Autolock _l(mLock);
- mTracks.add(record);
- }
- void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record)
- {
- Mutex::Autolock _l(mLock);
- destroyTrack_l(record);
- }
- void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config)
- {
- ThreadBase::getAudioPortConfig(config);
- config->role = AUDIO_PORT_ROLE_SINK;
- config->ext.mix.hw_module = mInput->audioHwDev->handle();
- config->ext.mix.usecase.source = mAudioSource;
- }
- // ----------------------------------------------------------------------------
- #ifdef QCOM_DIRECTTRACK
- AudioFlinger::DirectAudioTrack::DirectAudioTrack(const sp<AudioFlinger>& audioFlinger,
- int output, AudioSessionDescriptor *outputDesc,
- IDirectTrackClient* client, audio_output_flags_t outflag)
- : BnDirectTrack(), mIsPaused(false), mAudioFlinger(audioFlinger), mOutput(output), mOutputDesc(outputDesc),
- mClient(client), mEffectConfigChanged(false), mKillEffectsThread(false), mFlag(outflag),
- mEffectsThreadScratchBuffer(NULL)
- {
- if (mFlag & AUDIO_OUTPUT_FLAG_LPA) {
- ALOGV("create effects thread for LPA");
- createEffectThread();
- allocateBufPool();
- } else if (mFlag & AUDIO_OUTPUT_FLAG_TUNNEL) {
- ALOGV("create effects thread for TUNNEL");
- createEffectThread();
- }
- outputDesc->mVolumeScale = 1.0;
- mDeathRecipient = new PMDeathRecipient(this);
- acquireWakeLock();
- }
- void AudioFlinger::DirectAudioTrack::signalEffect() {
- if (mFlag & AUDIO_OUTPUT_FLAG_LPA){
- mEffectConfigChanged = true;
- mEffectCv.signal();
- }
- }
- AudioFlinger::DirectAudioTrack::~DirectAudioTrack() {
- if (mFlag & AUDIO_OUTPUT_FLAG_LPA) {
- requestAndWaitForEffectsThreadExit();
- mAudioFlinger->deleteEffectSession();
- deallocateBufPool();
- } else if (mFlag & AUDIO_OUTPUT_FLAG_TUNNEL) {
- requestAndWaitForEffectsThreadExit();
- mAudioFlinger->deleteEffectSession();
- }
- AudioSystem::releaseOutput(mOutput);
- releaseWakeLock();
- {
- Mutex::Autolock _l(pmLock);
- if (mPowerManager != 0) {
- sp<IBinder> binder = mPowerManager->asBinder();
- binder->unlinkToDeath(mDeathRecipient);
- }
- }
- }
- status_t AudioFlinger::DirectAudioTrack::start() {
- AudioSystem::startOutput(mOutput, (audio_stream_type_t)mOutputDesc->mStreamType, NULL);
- if(mIsPaused) {
- mIsPaused = false;
- mOutputDesc->stream->start(mOutputDesc->stream);
- }
- mOutputDesc->mActive = true;
- mOutputDesc->stream->set_volume(mOutputDesc->stream,
- mOutputDesc->mVolumeLeft * mOutputDesc->mVolumeScale,
- mOutputDesc->mVolumeRight* mOutputDesc->mVolumeScale);
- return NO_ERROR;
- }
- void AudioFlinger::DirectAudioTrack::stop() {
- ALOGV("DirectAudioTrack::stop");
- mOutputDesc->mActive = false;
- mOutputDesc->stream->stop(mOutputDesc->stream);
- AudioSystem::stopOutput(mOutput, (audio_stream_type_t)mOutputDesc->mStreamType, NULL);
- }
- void AudioFlinger::DirectAudioTrack::pause() {
- if(!mIsPaused) {
- mIsPaused = true;
- mOutputDesc->stream->pause(mOutputDesc->stream);
- mOutputDesc->mActive = false;
- AudioSystem::stopOutput(mOutput, (audio_stream_type_t)mOutputDesc->mStreamType,NULL);
- }
- }
- ssize_t AudioFlinger::DirectAudioTrack::write(const void *buffer, size_t size) {
- ALOGV("Writing to AudioSessionOut");
- int isAvail = 0;
- mOutputDesc->stream->is_buffer_available(mOutputDesc->stream, &isAvail);
- if (!isAvail) {
- return 0;
- }
- if (mFlag & AUDIO_OUTPUT_FLAG_LPA) {
- mEffectLock.lock();
- List<BufferInfo>::iterator it = mEffectsPool.begin();
- BufferInfo buf = *it;
- mEffectsPool.erase(it);
- memcpy((char *) buf.localBuf, (char *)buffer, size);
- buf.bytesToWrite = size;
- mEffectsPool.push_back(buf);
- mAudioFlinger->applyEffectsOn(static_cast<void *>(this),
- (int16_t*)buf.localBuf, (int16_t*)buffer, (int)size, true);
- mEffectLock.unlock();
- }
- ALOGV("out of Writing to AudioSessionOut");
- return mOutputDesc->stream->write(mOutputDesc->stream, buffer, size);
- }
- void AudioFlinger::DirectAudioTrack::flush() {
- if (mFlag & AUDIO_OUTPUT_FLAG_LPA) {
- mEffectLock.lock();
- mEffectsPool.clear();
- mEffectsPool = mBufPool;
- mEffectLock.unlock();
- }
- mOutputDesc->stream->flush(mOutputDesc->stream);
- }
- void AudioFlinger::DirectAudioTrack::mute(bool muted) {
- }
- void AudioFlinger::DirectAudioTrack::setVolume(float left, float right) {
- ALOGV("DirectAudioTrack::setVolume left: %f, right: %f", left, right);
- if(mOutputDesc) {
- mOutputDesc->mVolumeLeft = left;
- mOutputDesc->mVolumeRight = right;
- if(mOutputDesc->mActive && mOutputDesc->stream) {
- mOutputDesc->stream->set_volume(mOutputDesc->stream,
- left * mOutputDesc->mVolumeScale,
- right* mOutputDesc->mVolumeScale);
- } else {
- ALOGD("stream is not active, so cache and send when stream is active");
- }
- } else {
- ALOGD("output descriptor is not valid to set the volume");
- }
- }
- int64_t AudioFlinger::DirectAudioTrack::getTimeStamp() {
- int64_t time;
- mOutputDesc->stream->get_next_write_timestamp(mOutputDesc->stream, &time);
- ALOGV("Timestamp %lld",time);
- return time;
- }
- void AudioFlinger::DirectAudioTrack::postEOS(int64_t delayUs) {
- if (delayUs == 0 ) {
- ALOGV("Notify Audio Track of EOS event");
- mClient->notify(DIRECT_TRACK_EOS);
- } else {
- ALOGV("Notify Audio Track of hardware failure event");
- mClient->notify(DIRECT_TRACK_HW_FAIL);
- }
- }
- void AudioFlinger::DirectAudioTrack::allocateBufPool() {
- void *dsp_buf = NULL;
- void *local_buf = NULL;
- //1. get the ion buffer information
- struct buf_info* buf = NULL;
- mOutputDesc->stream->get_buffer_info(mOutputDesc->stream, &buf);
- ALOGV("get buffer info %p",buf);
- if (!buf) {
- ALOGV("buffer is NULL");
- return;
- }
- int nSize = buf->bufsize;
- int bufferCount = buf->nBufs;
- //2. allocate the buffer pool, allocate local buffers
- for (int i = 0; i < bufferCount; i++) {
- dsp_buf = (void *)buf->buffers[i];
- local_buf = malloc(nSize);
- memset(local_buf, 0, nSize);
- // Store this information for internal mapping / maintanence
- BufferInfo buf(local_buf, dsp_buf, nSize);
- buf.bytesToWrite = 0;
- mBufPool.push_back(buf);
- mEffectsPool.push_back(buf);
- ALOGV("The MEM that is allocated buffer is %x, size %d",(unsigned int)dsp_buf,nSize);
- }
- mEffectsThreadScratchBuffer = malloc(nSize);
- ALOGV("effectsThreadScratchBuffer = %x",mEffectsThreadScratchBuffer);
- free(buf);
- }
- void AudioFlinger::DirectAudioTrack::deallocateBufPool() {
- //1. Deallocate the local memory
- //2. Remove all the buffers from bufpool
- mEffectLock.lock();
- while (!mBufPool.empty()) {
- List<BufferInfo>::iterator it = mBufPool.begin();
- BufferInfo &memBuffer = *it;
- // free the local buffer corresponding to mem buffer
- if (memBuffer.localBuf) {
- free(memBuffer.localBuf);
- memBuffer.localBuf = NULL;
- }
- ALOGV("Removing from bufpool");
- mBufPool.erase(it);
- }
- mEffectsPool.clear();
- mEffectLock.unlock();
- free(mEffectsThreadScratchBuffer);
- mEffectsThreadScratchBuffer = NULL;
- }
- status_t AudioFlinger::DirectAudioTrack::onTransact(
- uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
- {
- return BnDirectTrack::onTransact(code, data, reply, flags);
- }
- AudioFlinger::DirectAudioTrack::AudioFlingerDirectTrackClient::AudioFlingerDirectTrackClient(void *obj)
- {
- ALOGV("AudioFlinger::DirectAudioTrack::AudioFlingerDirectTrackClient");
- pBaseClass = (DirectAudioTrack*)obj;
- }
- void AudioFlinger::DirectAudioTrack::AudioFlingerDirectTrackClient::binderDied(const wp<IBinder>& who) {
- pBaseClass->mAudioFlinger.clear();
- ALOGW("AudioFlinger server died!");
- }
- void AudioFlinger::DirectAudioTrack::AudioFlingerDirectTrackClient
- ::ioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2) {
- ALOGV("ioConfigChanged() event %d", event);
- if (event == AudioSystem::EFFECT_CONFIG_CHANGED) {
- ALOGV("Received notification for change in effect module");
- // Seek to current media time - flush the decoded buffers with the driver
- pBaseClass->mEffectConfigChanged = true;
- // Signal effects thread to re-apply effects
- ALOGV("Signalling Effects Thread");
- pBaseClass->mEffectCv.signal();
- }
- ALOGV("ioConfigChanged Out");
- }
- void AudioFlinger::DirectAudioTrack::acquireWakeLock()
- {
- Mutex::Autolock _l(pmLock);
- if (mPowerManager == 0) {
- // use checkService() to avoid blocking if power service is not up yet
- sp<IBinder> binder =
- defaultServiceManager()->checkService(String16("power"));
- if (binder == 0) {
- ALOGW("Thread %s cannot connect to the power manager service", lockName);
- } else {
- mPowerManager = interface_cast<IPowerManager>(binder);
- binder->linkToDeath(mDeathRecipient);
- }
- }
- if (mPowerManager != 0 && mWakeLockToken == 0) {
- sp<IBinder> binder = new BBinder();
- status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
- binder,
- String16(lockName),
- String16("media"));
- if (status == NO_ERROR) {
- mWakeLockToken = binder;
- }
- ALOGV("acquireWakeLock() status %d", status);
- }
- }
- void AudioFlinger::DirectAudioTrack::releaseWakeLock()
- {
- Mutex::Autolock _l(pmLock);
- if (mWakeLockToken != 0) {
- ALOGV("releaseWakeLock()");
- if (mPowerManager != 0) {
- mPowerManager->releaseWakeLock(mWakeLockToken, 0);
- }
- mWakeLockToken.clear();
- }
- }
- void AudioFlinger::DirectAudioTrack::clearPowerManager()
- {
- releaseWakeLock();
- Mutex::Autolock _l(pmLock);
- mPowerManager.clear();
- }
- void AudioFlinger::DirectAudioTrack::PMDeathRecipient::binderDied(const wp<IBinder>& who)
- {
- parentClass->clearPowerManager();
- ALOGW("power manager service died !!!");
- }
- #endif
- // ----------------------------------------------------------------------------
- }; // namespace android