/* ** ** Copyright 2007, The Android Open Source Project ** Copyright (c) 2011-2013, The Linux Foundation. All rights reserved. ** ** Not a Contribution, Apache license notifications and license are retained ** for attribution purposes only. ** ** 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 #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #undef ADD_BATTERY_DATA #ifdef ADD_BATTERY_DATA #include #include #endif #include #include #include #include #include "AudioMixer.h" #include "AudioFlinger.h" #include "ServiceUtilities.h" #include #include #include #include #include #include // #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds #ifdef DEBUG_CPU_USAGE #include #include #endif #include #include #include "FastMixer.h" // NBAIO implementations #include #include #include #include #include #include #include "SchedulingPolicyService.h" #ifdef SRS_PROCESSING #include "srs_processing.h" #include "postpro_patch_ics.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 #ifdef QCOM_HARDWARE #define DIRECT_TRACK_EOS 1 #define DIRECT_TRACK_HW_FAIL 6 static const char lockName[] = "DirectTrack"; #endif namespace android { static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; static const char kHardwareLockedString[] = "Hardware lock is taken\n"; static const float MAX_GAIN = 4096.0f; static const uint32_t MAX_GAIN_INT = 0x1000; // 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 = 5; static const int kDumpLockRetries = 50; static const int kDumpLockSleepUs = 20000; // 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 for setParameters to complete static const nsecs_t kSetParametersTimeoutNs = 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 mix buffer size, expressed in milliseconds rather than frames static const uint32_t kMinNormalMixBufferSizeMs = 20; // maximum normal mix buffer size static const uint32_t kMaxNormalMixBufferSizeMs = 24; nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; // 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; static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off" // AudioFlinger::setParameters() updates, other threads read w/o lock // Priorities for requestPriority static const int kPriorityAudioApp = 2; static const int kPriorityFastMixer = 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 double-buffering by default, but doesn't tell us about that. // So for now we just assume that client is double-buffered. // FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or // N-buffering, so AudioFlinger could allocate the right amount of memory. // See the client's minBufCount and mNotificationFramesAct calculations for details. static const int kFastTrackMultiplier = 2; // ---------------------------------------------------------------------------- #ifdef ADD_BATTERY_DATA // To collect the amplifier usage static void addBatteryData(uint32_t params) { sp service = IMediaDeathNotifier::getMediaPlayerService(); if (service == NULL) { // it already logged return; } service->addBatteryData(params); } #endif static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) { const hw_module_t *mod; int rc; rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); if (rc) { goto out; } rc = audio_hw_device_open(mod, dev); ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); if (rc) { goto out; } #if !defined(ICS_AUDIO_BLOB) && !defined(MR0_AUDIO_BLOB) if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); rc = BAD_VALUE; goto out; } #endif return 0; out: *dev = NULL; return rc; } static uint32_t getInputChannelCount(uint32_t channels) { #ifdef QCOM_HARDWARE // only mono or stereo and 5.1 are supported for input sources return popcount((channels) & (AUDIO_CHANNEL_IN_STEREO | AUDIO_CHANNEL_IN_MONO | AUDIO_CHANNEL_IN_5POINT1)); #else return popcount(channels); #endif } // ---------------------------------------------------------------------------- AudioFlinger::AudioFlinger() : BnAudioFlinger(), mPrimaryHardwareDev(NULL), mHardwareStatus(AUDIO_HW_IDLE), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), mMode(AUDIO_MODE_INVALID), mBtNrecIsOff(false) #ifdef QCOM_HARDWARE ,mAllChainsLocked(false) #endif { } void AudioFlinger::onFirstRef() { int rc = 0; #ifdef QCOM_HARDWARE mA2DPHandle = -1; #endif Mutex::Autolock _l(mLock); /* TODO: move all this work into an Init() function */ #ifdef QCOM_HARDWARE mLPASessionId = -2; // -2 is invalid session ID mIsEffectConfigChanged = false; mLPAEffectChain = NULL; #endif char val_str[PROPERTY_VALUE_MAX] = { 0 }; if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { uint32_t int_val; if (1 == sscanf(val_str, "%u", &int_val)) { mStandbyTimeInNsecs = milliseconds(int_val); ALOGI("Using %u mSec as standby time.", int_val); } else { mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; ALOGI("Using default %u mSec as standby time.", (uint32_t)(mStandbyTimeInNsecs / 1000000)); } } mMode = AUDIO_MODE_NORMAL; } AudioFlinger::~AudioFlinger() { while (!mRecordThreads.isEmpty()) { // closeInput_nonvirtual() will remove specified entry from mRecordThreads closeInput_nonvirtual(mRecordThreads.keyAt(0)); } while (!mPlaybackThreads.isEmpty()) { // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); } for (size_t i = 0; i < mAudioHwDevs.size(); i++) { // no mHardwareLock needed, as there are no other references to this audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); delete mAudioHwDevs.valueAt(i); } } static const char * const audio_interfaces[] = { AUDIO_HARDWARE_MODULE_ID_PRIMARY, AUDIO_HARDWARE_MODULE_ID_A2DP, AUDIO_HARDWARE_MODULE_ID_USB, }; #define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( audio_module_handle_t module, audio_devices_t devices) { // if module is 0, the request comes from an old policy manager and we should load // well known modules if (module == 0) { ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { loadHwModule_l(audio_interfaces[i]); } // then try to find a module supporting the requested device. for (size_t i = 0; i < mAudioHwDevs.size(); i++) { AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); audio_hw_device_t *dev = audioHwDevice->hwDevice(); if ((dev->get_supported_devices != NULL) && (dev->get_supported_devices(dev) & devices) == devices) return audioHwDevice; #ifdef ICS_AUDIO_BLOB else if (dev->get_supported_devices == NULL && i != 0 && devices == 0x80) // Reasonably safe assumption: A non-primary HAL without // get_supported_devices is a locally-built A2DP binary return audioHwDevice; #endif } } else { // check a match for the requested module handle AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); if (audioHwDevice != NULL) { return audioHwDevice; } } return NULL; } void AudioFlinger::dumpClients(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; result.append("Clients:\n"); for (size_t i = 0; i < mClients.size(); ++i) { sp client = mClients.valueAt(i).promote(); if (client != 0) { snprintf(buffer, SIZE, " pid: %d\n", client->pid()); result.append(buffer); } } result.append("Global session refs:\n"); result.append(" session pid count\n"); for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { AudioSessionRef *r = mAudioSessionRefs[i]; snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); result.append(buffer); } write(fd, result.string(), result.size()); } void AudioFlinger::dumpInternals(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; hardware_call_state hardwareStatus = mHardwareStatus; snprintf(buffer, SIZE, "Hardware status: %d\n" "Standby Time mSec: %u\n", hardwareStatus, (uint32_t)(mStandbyTimeInNsecs / 1000000)); result.append(buffer); write(fd, result.string(), result.size()); } void AudioFlinger::dumpPermissionDenial(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "Permission Denial: " "can't dump AudioFlinger from pid=%d, uid=%d\n", IPCThreadState::self()->getCallingPid(), IPCThreadState::self()->getCallingUid()); result.append(buffer); write(fd, result.string(), result.size()); } static bool tryLock(Mutex& mutex) { bool locked = false; for (int i = 0; i < kDumpLockRetries; ++i) { if (mutex.tryLock() == NO_ERROR) { locked = true; break; } usleep(kDumpLockSleepUs); } return locked; } status_t AudioFlinger::dump(int fd, const Vector& args) { if (!dumpAllowed()) { dumpPermissionDenial(fd, args); } else { // get state of hardware lock bool hardwareLocked = tryLock(mHardwareLock); if (!hardwareLocked) { String8 result(kHardwareLockedString); write(fd, result.string(), result.size()); } else { mHardwareLock.unlock(); } bool locked = tryLock(mLock); // failed to lock - AudioFlinger is probably deadlocked if (!locked) { String8 result(kDeadlockedString); write(fd, result.string(), result.size()); } dumpClients(fd, args); dumpInternals(fd, args); // dump playback threads for (size_t i = 0; i < mPlaybackThreads.size(); i++) { mPlaybackThreads.valueAt(i)->dump(fd, args); } // dump record threads for (size_t i = 0; i < mRecordThreads.size(); i++) { mRecordThreads.valueAt(i)->dump(fd, args); } // dump all hardware devs for (size_t i = 0; i < mAudioHwDevs.size(); i++) { audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); dev->dump(dev, fd); } if (locked) mLock.unlock(); } return NO_ERROR; } sp AudioFlinger::registerPid_l(pid_t pid) { // If pid is already in the mClients wp<> map, then use that entry // (for which promote() is always != 0), otherwise create a new entry and Client. sp client = mClients.valueFor(pid).promote(); if (client == 0) { client = new Client(this, pid); mClients.add(pid, client); } return client; } // IAudioFlinger interface sp AudioFlinger::createTrack( pid_t pid, audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, IAudioFlinger::track_flags_t flags, const sp& sharedBuffer, audio_io_handle_t output, pid_t tid, int *sessionId, status_t *status) { sp track; sp trackHandle; sp client; status_t lStatus; int lSessionId; // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, // but if someone uses binder directly they could bypass that and cause us to crash if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { ALOGE("createTrack() invalid stream type %d", streamType); lStatus = BAD_VALUE; goto Exit; } { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); PlaybackThread *effectThread = NULL; if (thread == NULL) { ALOGE("unknown output thread"); lStatus = BAD_VALUE; goto Exit; } client = registerPid_l(pid); ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { // check if an effect chain with the same session ID is present on another // output thread and move it here. for (size_t i = 0; i < mPlaybackThreads.size(); i++) { sp t = mPlaybackThreads.valueAt(i); if (mPlaybackThreads.keyAt(i) != output) { uint32_t sessions = t->hasAudioSession(*sessionId); if (sessions & PlaybackThread::EFFECT_SESSION) { effectThread = t.get(); break; } } } lSessionId = *sessionId; } else { // if no audio session id is provided, create one here lSessionId = nextUniqueId(); if (sessionId != NULL) { *sessionId = lSessionId; } } ALOGV("createTrack() lSessionId: %d", lSessionId); track = thread->createTrack_l(client, streamType, sampleRate, format, channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); // move effect chain to this output thread if an effect on same session was waiting // for a track to be created if (lStatus == NO_ERROR && effectThread != NULL) { Mutex::Autolock _dl(thread->mLock); Mutex::Autolock _sl(effectThread->mLock); moveEffectChain_l(lSessionId, effectThread, thread, true); } // Look for sync events awaiting for a session to be used. for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { if (lStatus == NO_ERROR) { (void) track->setSyncEvent(mPendingSyncEvents[i]); } else { mPendingSyncEvents[i]->cancel(); } mPendingSyncEvents.removeAt(i); i--; } } } } if (lStatus == NO_ERROR) { trackHandle = new TrackHandle(track); } else { // remove local strong reference to Client before deleting the Track so that the Client // destructor is called by the TrackBase destructor with mLock held client.clear(); track.clear(); } Exit: if (status != NULL) { *status = lStatus; } return trackHandle; } #ifdef QCOM_HARDWARE sp AudioFlinger::createDirectTrack( pid_t pid, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_io_handle_t output, int *sessionId, IDirectTrackClient *client, audio_stream_type_t streamType, status_t *status) { *status = NO_ERROR; status_t lStatus = NO_ERROR; sp track = NULL; DirectAudioTrack* directTrack = NULL; Mutex::Autolock _l(mLock); ALOGV("createDirectTrack() sessionId: %d sampleRate %d channelMask %d", *sessionId, sampleRate, channelMask); AudioSessionDescriptor *desc = mDirectAudioTracks.valueFor(output); if(desc == NULL) { ALOGE("Error: Invalid output (%d) to create direct audio track", output); lStatus = BAD_VALUE; goto Exit; } desc->mStreamType = streamType; if (desc->flag & AUDIO_OUTPUT_FLAG_LPA) { if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { for (size_t i = 0; i < mPlaybackThreads.size(); i++) { sp t = mPlaybackThreads.valueAt(i); // Check if the session ID is already associated with a track uint32_t sessions = t->hasAudioSession(*sessionId); // check if an effect with same session ID is waiting for a ssession to be created ALOGV("check if an effect with same session ID is waiting for a ssession to be created"); if ((mLPAEffectChain == NULL) && (sessions & PlaybackThread::EFFECT_SESSION)) { // Clear reference to previous effect chain if any t->mLock.lock(); ALOGV("getting the LPA effect chain and setting LPA flag to true."); mLPAEffectChain = t->getEffectChain_l(*sessionId); t->mLock.unlock(); } } mLPASessionId = *sessionId; if (mLPAEffectChain != NULL) { mLPAEffectChain->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 mLPAEffectChain->setVolume_l(&volume,&volume); } else { ALOGW("There was no effectChain created for the sessionId(%d)", mLPASessionId); } mLPASampleRate = sampleRate; mLPANumChannels = popcount(channelMask); } else { if(sessionId != NULL) { ALOGE("Error: Invalid sessionID (%d) for direct audio track", *sessionId); } } } mLock.unlock(); directTrack = new DirectAudioTrack(this, output, desc, client, desc->flag); desc->trackRefPtr = dynamic_cast(directTrack); mLock.lock(); if (directTrack != 0) { track = dynamic_cast(directTrack); AudioEventObserver* obv = dynamic_cast(directTrack); ALOGE("setting observer mOutputDesc track %p, obv %p", track.get(), obv); desc->stream->set_observer(desc->stream, reinterpret_cast(obv)); } else { lStatus = BAD_VALUE; } Exit: if(lStatus) { if (track != NULL) { track.clear(); } *status = lStatus; } return track; } void AudioFlinger::deleteEffectSession() { ALOGV("deleteSession"); // -2 is invalid session ID mLPASessionId = -2; if (mLPAEffectChain != NULL) { mLPAEffectChain->lock(); mLPAEffectChain->setLPAFlag(false); size_t i, numEffects = mLPAEffectChain->getNumEffects(); for(i = 0; i < numEffects; i++) { sp effect = mLPAEffectChain->getEffectFromIndex_l(i); effect->setInBuffer(mLPAEffectChain->inBuffer()); if (i == numEffects-1) { effect->setOutBuffer(mLPAEffectChain->outBuffer()); } else { effect->setOutBuffer(mLPAEffectChain->inBuffer()); } effect->configure(); } mLPAEffectChain->unlock(); mLPAEffectChain.clear(); mLPAEffectChain = NULL; } } // ToDo: Should we go ahead with this frameCount? #define DEAFULT_FRAME_COUNT 1200 bool AudioFlinger::applyEffectsOn(void *token, int16_t *inBuffer, int16_t *outBuffer, int size, bool force) { ALOGV("applyEffectsOn: inBuf %p outBuf %p size %d token %p", inBuffer, outBuffer, size, token); // This might be the first buffer to apply effects after effect config change // should not skip effects processing mIsEffectConfigChanged = false; volatile size_t numEffects = 0; #ifdef SRS_PROCESSING POSTPRO_PATCH_ICS_OUTPROC_DIRECT_SAMPLES(token, AUDIO_FORMAT_PCM_16_BIT, outBuffer, size, mLPASampleRate, mLPANumChannels); #endif if(mLPAEffectChain != NULL) { numEffects = mLPAEffectChain->getNumEffects(); } if( numEffects > 0) { size_t i = 0; int16_t *pIn = inBuffer; int16_t *pOut = outBuffer; int frameCount = size / (sizeof(int16_t) * mLPANumChannels); while(frameCount > 0) { if(mLPAEffectChain == NULL) { ALOGV("LPA Effect Chain is removed - No effects processing !!"); numEffects = 0; break; } mLPAEffectChain->lock(); numEffects = mLPAEffectChain->getNumEffects(); if(!numEffects) { ALOGV("applyEffectsOn: All the effects are removed - nothing to process"); mLPAEffectChain->unlock(); break; } int outFrameCount = (frameCount > DEAFULT_FRAME_COUNT ? DEAFULT_FRAME_COUNT: frameCount); bool isEffectEnabled = false; for(i = 0; i < numEffects; i++) { // If effect configuration is changed while applying effects do not process further if(mIsEffectConfigChanged && !force) { mLPAEffectChain->unlock(); ALOGV("applyEffectsOn: mIsEffectConfigChanged is set - no further processing %d",frameCount); return false; } sp effect = mLPAEffectChain->getEffectFromIndex_l(i); if(effect == NULL) { ALOGE("getEffectFromIndex_l(%d) returned NULL ptr", i); mLPAEffectChain->unlock(); return false; } if(i == 0) { // For the first set input and output buffers different isEffectEnabled = effect->isProcessEnabled(); effect->setInBuffer(pIn); effect->setOutBuffer(pOut); } else { // For the remaining use previous effect's output buffer as input buffer effect->setInBuffer(pOut); effect->setOutBuffer(pOut); } // true indicates that it is being applied on LPA output effect->configure(true, mLPASampleRate, mLPANumChannels, outFrameCount); } if(isEffectEnabled) { // Clear the output buffer memset(pOut, 0, (outFrameCount * mLPANumChannels * sizeof(int16_t))); } else { // Copy input buffer content to the output buffer memcpy(pOut, pIn, (outFrameCount * mLPANumChannels * sizeof(int16_t))); } mLPAEffectChain->process_l(); mLPAEffectChain->unlock(); // Update input and output buffer pointers pIn += (outFrameCount * mLPANumChannels); pOut += (outFrameCount * mLPANumChannels); frameCount -= outFrameCount; } } if (!numEffects && !force) { ALOGV("applyEffectsOn: There are no effects to be applied"); if(inBuffer != outBuffer) { // No effect applied so just copy input buffer to output buffer memcpy(outBuffer, inBuffer, size); } } return true; } #endif uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const { Mutex::Autolock _l(mLock); #ifdef QCOM_HARDWARE if (!mDirectAudioTracks.isEmpty()) { AudioSessionDescriptor *desc = mDirectAudioTracks.valueFor(output); if(desc != NULL) { return desc->stream->common.get_sample_rate(&desc->stream->common); } } #endif PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { ALOGW("sampleRate() unknown thread %d", output); return 0; } return thread->sampleRate(); } int AudioFlinger::channelCount(audio_io_handle_t output) const { Mutex::Autolock _l(mLock); #ifdef QCOM_HARDWARE AudioSessionDescriptor *desc = mDirectAudioTracks.valueFor(output); if(desc != NULL) { return desc->stream->common.get_channels(&desc->stream->common); } #endif PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { ALOGW("channelCount() unknown thread %d", output); return 0; } return thread->channelCount(); } audio_format_t AudioFlinger::format(audio_io_handle_t output) const { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { ALOGW("format() unknown thread %d", output); return AUDIO_FORMAT_INVALID; } return thread->format(); } size_t AudioFlinger::frameCount(audio_io_handle_t output) const { Mutex::Autolock _l(mLock); #ifdef QCOM_HARDWARE AudioSessionDescriptor *desc = mDirectAudioTracks.valueFor(output); if(desc != NULL) { return desc->stream->common.get_buffer_size(&desc->stream->common); } #endif PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { ALOGW("frameCount() unknown thread %d", output); return 0; } // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; // should examine all callers and fix them to handle smaller counts return thread->frameCount(); } uint32_t AudioFlinger::latency(audio_io_handle_t output) const { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { ALOGW("latency() unknown thread %d", output); return 0; } return thread->latency(); } status_t AudioFlinger::setMasterVolume(float value) { status_t ret = initCheck(); if (ret != NO_ERROR) { return ret; } // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } #ifdef QCOM_HARDWARE mA2DPHandle = -1; #endif Mutex::Autolock _l(mLock); mMasterVolume = value; // Set master volume in the HALs which support it. for (size_t i = 0; i < mAudioHwDevs.size(); i++) { AutoMutex lock(mHardwareLock); AudioHwDevice *dev = mAudioHwDevs.valueAt(i); mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; if (dev->canSetMasterVolume()) { dev->hwDevice()->set_master_volume(dev->hwDevice(), value); } mHardwareStatus = AUDIO_HW_IDLE; } // Now set the master volume in each playback thread. Playback threads // assigned to HALs which do not have master volume support will apply // master volume during the mix operation. Threads with HALs which do // support master volume will simply ignore the setting. for (size_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->isDuplicating()) { continue; } mPlaybackThreads.valueAt(i)->setMasterVolume(value); } return NO_ERROR; } status_t AudioFlinger::setMode(audio_mode_t mode) { status_t ret = initCheck(); if (ret != NO_ERROR) { return ret; } // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } if (uint32_t(mode) >= AUDIO_MODE_CNT) { ALOGW("Illegal value: setMode(%d)", mode); return BAD_VALUE; } { // scope for the lock AutoMutex lock(mHardwareLock); audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); mHardwareStatus = AUDIO_HW_SET_MODE; ret = dev->set_mode(dev, mode); mHardwareStatus = AUDIO_HW_IDLE; } if (NO_ERROR == ret) { Mutex::Autolock _l(mLock); mMode = mode; for (size_t i = 0; i < mPlaybackThreads.size(); i++) mPlaybackThreads.valueAt(i)->setMode(mode); } return ret; } status_t AudioFlinger::setMicMute(bool state) { status_t ret = initCheck(); if (ret != NO_ERROR) { return ret; } // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } AutoMutex lock(mHardwareLock); audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; ret = dev->set_mic_mute(dev, state); mHardwareStatus = AUDIO_HW_IDLE; return ret; } bool AudioFlinger::getMicMute() const { status_t ret = initCheck(); if (ret != NO_ERROR) { return false; } bool state = AUDIO_MODE_INVALID; AutoMutex lock(mHardwareLock); audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; dev->get_mic_mute(dev, &state); mHardwareStatus = AUDIO_HW_IDLE; return state; } status_t AudioFlinger::setMasterMute(bool muted) { status_t ret = initCheck(); if (ret != NO_ERROR) { return ret; } // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } Mutex::Autolock _l(mLock); mMasterMute = muted; // Set master mute in the HALs which support it. #ifndef ICS_AUDIO_BLOB for (size_t i = 0; i < mAudioHwDevs.size(); i++) { AutoMutex lock(mHardwareLock); AudioHwDevice *dev = mAudioHwDevs.valueAt(i); mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; if (dev->canSetMasterMute()) { dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); } mHardwareStatus = AUDIO_HW_IDLE; } #endif // Now set the master mute in each playback thread. Playback threads // assigned to HALs which do not have master mute support will apply master // mute during the mix operation. Threads with HALs which do support master // mute will simply ignore the setting. for (size_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->isDuplicating()) { continue; } mPlaybackThreads.valueAt(i)->setMasterMute(muted); } return NO_ERROR; } float AudioFlinger::masterVolume() const { Mutex::Autolock _l(mLock); return masterVolume_l(); } bool AudioFlinger::masterMute() const { Mutex::Autolock _l(mLock); return masterMute_l(); } float AudioFlinger::masterVolume_l() const { return mMasterVolume; } bool AudioFlinger::masterMute_l() const { return mMasterMute; } status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, audio_io_handle_t output) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } if (uint32_t(stream) >= AUDIO_STREAM_CNT) { ALOGE("setStreamVolume() invalid stream %d", stream); return BAD_VALUE; } AutoMutex lock(mLock); #ifdef QCOM_HARDWARE ALOGV("setStreamVolume stream %d, output %d, value %f",stream, output, value); AudioSessionDescriptor *desc = NULL; if (!mDirectAudioTracks.isEmpty()) { desc = mDirectAudioTracks.valueFor(output); if (desc != NULL) { ALOGV("setStreamVolume for mAudioTracks size %d desc %p",mDirectAudioTracks.size(),desc); if (desc->mStreamType == stream) { mStreamTypes[stream].volume = value; desc->mVolumeScale = value; desc->stream->set_volume(desc->stream, desc->mVolumeLeft * mStreamTypes[stream].volume, desc->mVolumeRight* mStreamTypes[stream].volume); return NO_ERROR; } } } #endif PlaybackThread *thread = NULL; if (output) { thread = checkPlaybackThread_l(output); if (thread == NULL) { #ifdef QCOM_HARDWARE if (desc != NULL) { return NO_ERROR; } #endif return BAD_VALUE; } } mStreamTypes[stream].volume = value; if (thread == NULL) { for (size_t i = 0; i < mPlaybackThreads.size(); i++) { mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); } } else { thread->setStreamVolume(stream, value); } return NO_ERROR; } status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } if (uint32_t(stream) >= AUDIO_STREAM_CNT || uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { ALOGE("setStreamMute() invalid stream %d", stream); return BAD_VALUE; } AutoMutex lock(mLock); mStreamTypes[stream].mute = muted; for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); return NO_ERROR; } float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const { if (uint32_t(stream) >= AUDIO_STREAM_CNT) { return 0.0f; } AutoMutex lock(mLock); float volume; if (output) { PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { return 0.0f; } volume = thread->streamVolume(stream); } else { volume = streamVolume_l(stream); } return volume; } bool AudioFlinger::streamMute(audio_stream_type_t stream) const { if (uint32_t(stream) >= AUDIO_STREAM_CNT) { return true; } AutoMutex lock(mLock); return streamMute_l(stream); } status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) { ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } // ioHandle == 0 means the parameters are global to the audio hardware interface if (ioHandle == 0) { Mutex::Autolock _l(mLock); #ifdef SRS_PROCESSING POSTPRO_PATCH_ICS_PARAMS_SET(keyValuePairs); if (!mDirectAudioTracks.isEmpty()) audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL); #endif status_t final_result = NO_ERROR; { AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_SET_PARAMETER; for (size_t i = 0; i < mAudioHwDevs.size(); i++) { audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); status_t result = dev->set_parameters(dev, keyValuePairs.string()); final_result = result ?: final_result; } mHardwareStatus = AUDIO_HW_IDLE; } #ifdef QCOM_HARDWARE AudioParameter param = AudioParameter(keyValuePairs); String8 value, key; int i = 0; key = String8(AudioParameter::keyADSPStatus); if (param.get(key, value) == NO_ERROR) { ALOGV("Set keyADSPStatus:%s", value.string()); if (value == "ONLINE" || value == "OFFLINE") { if (!mDirectAudioTracks.isEmpty()) { for (i=0; i < mDirectAudioTracks.size(); i++) { mDirectAudioTracks.valueAt(i)->stream->common.set_parameters( &mDirectAudioTracks.valueAt(i)->stream->common, keyValuePairs.string()); } } } } #else // disable AEC and NS if the device is a BT SCO headset supporting those pre processings AudioParameter param = AudioParameter(keyValuePairs); String8 value; #endif if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); if (mBtNrecIsOff != btNrecIsOff) { for (size_t i = 0; i < mRecordThreads.size(); i++) { sp thread = mRecordThreads.valueAt(i); audio_devices_t device = thread->inDevice(); bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; // collect all of the thread's session IDs KeyedVector ids = thread->sessionIds(); // suspend effects associated with those session IDs for (size_t j = 0; j < ids.size(); ++j) { int sessionId = ids.keyAt(j); thread->setEffectSuspended(FX_IID_AEC, suspend, sessionId); thread->setEffectSuspended(FX_IID_NS, suspend, sessionId); } } mBtNrecIsOff = btNrecIsOff; } } String8 screenState; if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { bool isOff = screenState == "off"; if (isOff != (gScreenState & 1)) { gScreenState = ((gScreenState & ~1) + 2) | isOff; } } return final_result; } #ifdef QCOM_HARDWARE AudioSessionDescriptor *desc = NULL; if (!mDirectAudioTracks.isEmpty()) { desc = mDirectAudioTracks.valueFor(ioHandle); if (desc != NULL) { ALOGV("setParameters for mAudioTracks size %d desc %p",mDirectAudioTracks.size(),desc); desc->stream->common.set_parameters(&desc->stream->common, keyValuePairs.string()); AudioParameter param = AudioParameter(keyValuePairs); String8 key = String8(AudioParameter::keyRouting); int device; if (param.getInt(key, device) == NO_ERROR) { #ifdef SRS_PROCESSING ALOGV("setParameters:: routing change to device %d", device); desc->device = (audio_devices_t)device; POSTPRO_PATCH_ICS_OUTPROC_MIX_ROUTE(desc->trackRefPtr, param, device); #endif if(mLPAEffectChain != NULL){ mLPAEffectChain->setDevice_l(device); audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL); } } } } #endif // hold a strong ref on thread in case closeOutput() or closeInput() is called // and the thread is exited once the lock is released sp thread; { Mutex::Autolock _l(mLock); thread = checkPlaybackThread_l(ioHandle); if (thread == 0) { thread = checkRecordThread_l(ioHandle); } else if (thread == primaryPlaybackThread_l()) { // indicate output device change to all input threads for pre processing AudioParameter param = AudioParameter(keyValuePairs); int value; DefaultKeyedVector< int, sp > recordThreads = mRecordThreads; mLock.unlock(); if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && (value != 0)) { for (size_t i = 0; i < recordThreads.size(); i++) { recordThreads.valueAt(i)->setParameters(keyValuePairs); } } mLock.lock(); } mLock.unlock(); } if (thread != 0) { return thread->setParameters(keyValuePairs); } return BAD_VALUE; } String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const { // ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", // ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); Mutex::Autolock _l(mLock); if (ioHandle == 0) { String8 out_s8; #ifdef SRS_PROCESSING POSTPRO_PATCH_ICS_PARAMS_GET(keys, out_s8); #endif for (size_t i = 0; i < mAudioHwDevs.size(); i++) { char *s; { AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_GET_PARAMETER; audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); s = dev->get_parameters(dev, keys.string()); mHardwareStatus = AUDIO_HW_IDLE; } out_s8 += String8(s ? s : ""); free(s); } return out_s8; } PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); if (playbackThread != NULL) { return playbackThread->getParameters(keys); } RecordThread *recordThread = checkRecordThread_l(ioHandle); if (recordThread != NULL) { return recordThread->getParameters(keys); } return String8(""); } size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask) const { status_t ret = initCheck(); if (ret != NO_ERROR) { return 0; } AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; struct audio_config config = { sample_rate: sampleRate, channel_mask: channelMask, format: format, }; audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); #ifndef ICS_AUDIO_BLOB size_t size = dev->get_input_buffer_size(dev, &config); #else size_t size = dev->get_input_buffer_size(dev, sampleRate, format, popcount(channelMask)); #endif mHardwareStatus = AUDIO_HW_IDLE; return size; } unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const { Mutex::Autolock _l(mLock); RecordThread *recordThread = checkRecordThread_l(ioHandle); if (recordThread != NULL) { return recordThread->getInputFramesLost(); } return 0; } status_t AudioFlinger::setVoiceVolume(float value) { status_t ret = initCheck(); if (ret != NO_ERROR) { return ret; } // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } AutoMutex lock(mHardwareLock); audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; ret = dev->set_voice_volume(dev, value); mHardwareStatus = AUDIO_HW_IDLE; return ret; } status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, audio_io_handle_t output) const { status_t status; Mutex::Autolock _l(mLock); PlaybackThread *playbackThread = checkPlaybackThread_l(output); if (playbackThread != NULL) { return playbackThread->getRenderPosition(halFrames, dspFrames); } return BAD_VALUE; } #ifdef QCOM_FM_ENABLED status_t AudioFlinger::setFmVolume(float value) { status_t ret = initCheck(); if (ret != NO_ERROR) { return ret; } // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } AutoMutex lock(mHardwareLock); audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); mHardwareStatus = AUDIO_SET_FM_VOLUME; ret = dev->set_fm_volume(dev, value); mHardwareStatus = AUDIO_HW_IDLE; return ret; } #endif void AudioFlinger::registerClient(const sp& client) { Mutex::Autolock _l(mLock); sp binder = client->asBinder(); if (mNotificationClients.indexOfKey(binder) < 0) { sp notificationClient = new NotificationClient(this, client, binder); ALOGV("registerClient() client %p, binder %d", notificationClient.get(), binder.get()); mNotificationClients.add(binder, notificationClient); sp binder = client->asBinder(); binder->linkToDeath(notificationClient); // the config change is always sent from playback or record threads to avoid deadlock // with AudioSystem::gLock for (size_t i = 0; i < mPlaybackThreads.size(); i++) { mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); } for (size_t i = 0; i < mRecordThreads.size(); i++) { mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); } } #ifdef QCOM_HARDWARE // Send the notification to the client only once. if (mA2DPHandle != -1) { ALOGV("A2DP active. Notifying the registered client"); client->ioConfigChanged(AudioSystem::A2DP_OUTPUT_STATE, mA2DPHandle, &mA2DPHandle); } #endif } #ifdef QCOM_HARDWARE status_t AudioFlinger::deregisterClient(const sp& client) { ALOGV("deregisterClient() %p, tid %d, calling tid %d", client.get(), gettid(), IPCThreadState::self()->getCallingPid()); Mutex::Autolock _l(mLock); sp binder = client->asBinder(); int index = mNotificationClients.indexOfKey(binder); if (index >= 0) { mNotificationClients.removeItemsAt(index); return true; } return false; } #endif void AudioFlinger::removeNotificationClient(sp binder) { Mutex::Autolock _l(mLock); mNotificationClients.removeItem(binder); int pid = IPCThreadState::self()->getCallingPid(); ALOGV("%d died, releasing its sessions", pid); size_t num = mAudioSessionRefs.size(); bool removed = false; for (size_t i = 0; i< num; ) { AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); ALOGV(" pid %d @ %d", ref->mPid, i); if (ref->mPid == pid) { ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); mAudioSessionRefs.removeAt(i); delete ref; removed = true; num--; } else { i++; } } if (removed) { purgeStaleEffects_l(); } } // audioConfigChanged_l() must be called with AudioFlinger::mLock held void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) { #ifdef QCOM_HARDWARE ALOGV("AudioFlinger::audioConfigChanged_l: event %d", event); if (event == AudioSystem::EFFECT_CONFIG_CHANGED) { mIsEffectConfigChanged = true; } #endif size_t size = mNotificationClients.size(); for (size_t i = 0; i < size; i++) { mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, param2); } } // removeClient_l() must be called with AudioFlinger::mLock held void AudioFlinger::removeClient_l(pid_t pid) { ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); mClients.removeItem(pid); } // getEffectThread_l() must be called with AudioFlinger::mLock held sp AudioFlinger::getEffectThread_l(int sessionId, int EffectId) { sp thread; for (size_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { ALOG_ASSERT(thread == 0); thread = mPlaybackThreads.valueAt(i); } } return thread; } // ---------------------------------------------------------------------------- AudioFlinger::ThreadBase::ThreadBase(const sp& 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(0), mFrameCount(0), mNormalFrameCount(0), // mChannelMask mChannelCount(0), mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), mParamStatus(NO_ERROR), 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() { mParamCond.broadcast(); // do not lock the mutex in destructor releaseWakeLock_l(); if (mPowerManager != 0) { sp binder = mPowerManager->asBinder(); binder->unlinkToDeath(mDeathRecipient); } } 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); mNewParameters.add(keyValuePairs); mWaitWorkCV.signal(); // wait condition with timeout in case the thread loop has exited // before the request could be processed if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { status = mParamStatus; mWaitWorkCV.signal(); } else { status = TIMED_OUT; } return status; } #ifdef QCOM_HARDWARE void AudioFlinger::ThreadBase::effectConfigChanged() { ALOGV("New effect is being added to LPA chain, Notifying LPA Direct Track"); mAudioFlinger->audioConfigChanged_l(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) { IoConfigEvent *ioEvent = new IoConfigEvent(event, param); mConfigEvents.add(static_cast(ioEvent)); ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); mWaitWorkCV.signal(); } // sendPrioConfigEvent_l() must be called with ThreadBase::mLock held void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) { PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); mConfigEvents.add(static_cast(prioEvent)); ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", mConfigEvents.size(), pid, tid, prio); mWaitWorkCV.signal(); } void AudioFlinger::ThreadBase::processConfigEvents() { mLock.lock(); while (!mConfigEvents.isEmpty()) { ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); ConfigEvent *event = mConfigEvents[0]; mConfigEvents.removeAt(0); // release mLock before locking AudioFlinger mLock: lock order is always // AudioFlinger then ThreadBase to avoid cross deadlock mLock.unlock(); switch(event->type()) { case CFG_EVENT_PRIO: { PrioConfigEvent *prioEvent = static_cast(event); int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); if (err != 0) { ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); } } break; case CFG_EVENT_IO: { IoConfigEvent *ioEvent = static_cast(event); mAudioFlinger->mLock.lock(); audioConfigChanged_l(ioEvent->event(), ioEvent->param()); mAudioFlinger->mLock.unlock(); } break; default: ALOGE("processConfigEvents() unknown event type %d", event->type()); break; } delete event; mLock.lock(); } mLock.unlock(); } void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; bool locked = tryLock(mLock); if (!locked) { snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); write(fd, buffer, strlen(buffer)); } snprintf(buffer, SIZE, "io handle: %d\n", mId); result.append(buffer); snprintf(buffer, SIZE, "TID: %d\n", getTid()); result.append(buffer); snprintf(buffer, SIZE, "standby: %d\n", mStandby); result.append(buffer); snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); result.append(buffer); snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); result.append(buffer); snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); result.append(buffer); snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); result.append(buffer); snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); result.append(buffer); snprintf(buffer, SIZE, "Format: %d\n", mFormat); result.append(buffer); snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); result.append(buffer); snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); result.append(buffer); result.append(" Index Command"); for (size_t i = 0; i < mNewParameters.size(); ++i) { snprintf(buffer, SIZE, "\n %02d ", i); result.append(buffer); result.append(mNewParameters[i]); } snprintf(buffer, SIZE, "\n\nPending config events: \n"); result.append(buffer); for (size_t i = 0; i < mConfigEvents.size(); i++) { mConfigEvents[i]->dump(buffer, SIZE); result.append(buffer); } result.append("\n"); write(fd, result.string(), result.size()); if (locked) { mLock.unlock(); } } void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); write(fd, buffer, strlen(buffer)); for (size_t i = 0; i < mEffectChains.size(); ++i) { sp chain = mEffectChains[i]; if (chain != 0) { chain->dump(fd, args); } } } void AudioFlinger::ThreadBase::acquireWakeLock() { Mutex::Autolock _l(mLock); acquireWakeLock_l(); } void AudioFlinger::ThreadBase::acquireWakeLock_l() { if (mPowerManager == 0) { // use checkService() to avoid blocking if power service is not up yet sp binder = defaultServiceManager()->checkService(String16("power")); if (binder == 0) { ALOGW("Thread %s cannot connect to the power manager service", mName); } else { mPowerManager = interface_cast(binder); binder->linkToDeath(mDeathRecipient); } } if (mPowerManager != 0) { sp binder = new BBinder(); status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, binder, String16(mName)); 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); } mWakeLockToken.clear(); } } void AudioFlinger::ThreadBase::clearPowerManager() { Mutex::Autolock _l(mLock); releaseWakeLock_l(); mPowerManager.clear(); } void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp& who) { sp 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 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& chain) { ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); if (index < 0) { return; } const KeyedVector >& sessionEffects = mSuspendedSessions.valueAt(index); for (size_t i = 0; i < sessionEffects.size(); i++) { sp 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 > 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 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& effect, bool enabled, int sessionId) { Mutex::Autolock _l(mLock); checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); } void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp& 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 chain = getEffectChain_l(sessionId); if (chain != 0) { chain->checkSuspendOnEffectEnabled(effect, enabled); } } // ---------------------------------------------------------------------------- AudioFlinger::PlaybackThread::PlaybackThread(const sp& audioFlinger, AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, type_t type) : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), mMixBuffer(NULL), mSuspended(0), mBytesWritten(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), mScreenState(gScreenState), // index 0 is reserved for normal mixer's submix mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) { snprintf(mName, kNameLength, "AudioOut_%X", id); // 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(); // 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_type_t) 0; 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() { delete [] mMixBuffer; } void AudioFlinger::PlaybackThread::dump(int fd, const Vector& args) { dumpInternals(fd, args); dumpTracks(fd, args); dumpEffectChains(fd, args); } void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 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(); snprintf(buffer, SIZE, "Output thread %p tracks\n", this); result.append(buffer); Track::appendDumpHeader(result); for (size_t i = 0; i < mTracks.size(); ++i) { sp track = mTracks[i]; if (track != 0) { track->dump(buffer, SIZE); result.append(buffer); } } snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); result.append(buffer); Track::appendDumpHeader(result); for (size_t i = 0; i < mActiveTracks.size(); ++i) { sp track = mActiveTracks[i].promote(); if (track != 0) { track->dump(buffer, SIZE); result.append(buffer); } } write(fd, result.string(), result.size()); // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. FastTrackUnderruns underruns = getFastTrackUnderruns(0); fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); } void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); result.append(buffer); snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); result.append(buffer); snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); result.append(buffer); snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); result.append(buffer); snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); result.append(buffer); snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); result.append(buffer); snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); result.append(buffer); write(fd, result.string(), result.size()); fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); dumpBase(fd, args); } // Thread virtuals status_t AudioFlinger::PlaybackThread::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::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::createTrack_l( const sp& client, audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, const sp& sharedBuffer, int sessionId, IAudioFlinger::track_flags_t flags, pid_t tid, status_t *status) { sp 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 >= (int) (mFrameCount * kFastTrackMultiplier))) ) ) && // PCM data audio_is_linear_pcm(format) && // mono or stereo ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && #ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE // hardware sample rate (sampleRate == mSampleRate) && #endif // 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) { frameCount = mFrameCount * kFastTrackMultiplier; } 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=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d " "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 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 = 0; if(mSampleRate) minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); if (minBufCount < 2) { minBufCount = 2; } int minFrameCount = mNormalFrameCount * minBufCount; if (frameCount < minFrameCount) { frameCount = minFrameCount; } } } if (mType == DIRECT) { #ifdef QCOM_ENHANCED_AUDIO if (((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) ||((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AMR_NB) ||((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AMR_WB) ||((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_EVRC) ||((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_EVRCB) ||((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_EVRCWB)) #else if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) #endif { if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" "for output %p with format %d", sampleRate, format, channelMask, mOutput, mFormat); lStatus = BAD_VALUE; goto Exit; } } } else { // Resampler implementation limits input sampling rate to 2 x output sampling rate. if (sampleRate > mSampleRate*2) { ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); lStatus = BAD_VALUE; goto Exit; } } lStatus = initCheck(); if (lStatus != NO_ERROR) { ALOGE("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 t = mTracks[i]; if (t != 0 && !t->isOutputTrack()) { 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, sharedBuffer, sessionId, flags); } else { track = TimedTrack::create(this, client, streamType, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId); } if (track == 0 || track->getCblk() == NULL || track->name() < 0) { lStatus = NO_MEMORY; goto Exit; } mTracks.add(track); sp 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: if (status) { *status = lStatus; } return track; } uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const { if (mFastMixer != NULL) { MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); if(mSampleRate) latency += (pipe->getAvgFrames() * 1000) / mSampleRate; else ALOGW("SampleRate is 0"); } return latency; } uint32_t AudioFlinger::PlaybackThread::correctLatency(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(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; } void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) { Mutex::Autolock _l(mLock); mStreamTypes[stream].mute = muted; } 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) { 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. track->mFillingUpStatus = Track::FS_FILLING; track->mResetDone = false; track->mPresentationCompleteFrames = 0; mActiveTracks.add(track); if (track->mainBuffer() != mMixBuffer) { sp 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; } ALOGV("mWaitWorkCV.broadcast"); mWaitWorkCV.broadcast(); return status; } // destroyTrack_l() must be called with ThreadBase::mLock held void AudioFlinger::PlaybackThread::destroyTrack_l(const sp& track) { track->mState = TrackBase::TERMINATED; // active tracks are removed by threadLoop() if (mActiveTracks.indexOf(track) < 0) { removeTrack_l(track); } } void AudioFlinger::PlaybackThread::removeTrack_l(const sp& 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 chain = getEffectChain_l(track->sessionId()); if (chain != 0) { chain->decTrackCnt(); } } String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) { String8 out_s8 = String8(""); char *s; Mutex::Autolock _l(mLock); if (initCheck() != NO_ERROR) { return out_s8; } s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); out_s8 = String8(s); free(s); return out_s8; } // audioConfigChanged_l() must be called with AudioFlinger::mLock held void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { AudioSystem::OutputDescriptor desc; void *param2 = NULL; ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); switch (event) { case AudioSystem::OUTPUT_OPENED: case AudioSystem::OUTPUT_CONFIG_CHANGED: desc.channels = mChannelMask; desc.samplingRate = mSampleRate; desc.format = mFormat; desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) desc.latency = latency(); param2 = &desc; break; case AudioSystem::STREAM_CONFIG_CHANGED: param2 = ¶m; case AudioSystem::OUTPUT_CLOSED: default: break; } mAudioFlinger->audioConfigChanged_l(event, mId, param2); } void AudioFlinger::PlaybackThread::readOutputParameters() { mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); mChannelCount = (uint16_t)popcount(mChannelMask); mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); mFrameSize = audio_stream_frame_size(&mOutput->stream->common); mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; if (mFrameCount & 15) { ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", mFrameCount); } // Calculate size of normal mix buffer relative to the HAL output buffer size double multiplier = 1.0; if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * 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 mix 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 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); delete[] mMixBuffer; mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); // 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() 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 > 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; } *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); if (isSuspended()) { // return an estimation of rendered frames when the output is suspended int32_t frames = mBytesWritten - latency_l(); if (frames < 0) { frames = 0; } *dspFrames = (uint32_t)frames; return NO_ERROR; } else { return mOutput->stream->get_render_position(mOutput->stream, dspFrames); } } 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 = mTracks[i]; if (sessionId == track->sessionId() && !(track->mCblk->flags & CBLK_INVALID_MSK)) { 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 = mTracks[i]; if (sessionId == track->sessionId() && !(track->mCblk->flags & CBLK_INVALID_MSK)) { 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& event) { if (!isValidSyncEvent(event)) { return BAD_VALUE; } Mutex::Autolock _l(mLock); for (size_t i = 0; i < mTracks.size(); ++i) { sp 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& event) const { return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; } void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp >& tracksToRemove) { size_t count = tracksToRemove.size(); if (CC_UNLIKELY(count)) { for (size_t i = 0 ; i < count ; i++) { const sp& track = tracksToRemove.itemAt(i); if ((track->sharedBuffer() != 0) && (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); } } } } // ---------------------------------------------------------------------------- AudioFlinger::MixerThread::MixerThread(const sp& 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=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " "mFrameCount=%d, mNormalFrameCount=%d", mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, mNormalFrameCount); mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); // FIXME - Current mixer implementation only supports stereo output if (mChannelCount != FCC_2) { ALOGE("Invalid audio hardware channel count %d", mChannelCount); } // 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)}; 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) { // create a MonoPipe to connect our submix to FastMixer NBAIO_Format format = mOutputSink->format(); // 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_FRAMES // create a Pipe to archive a copy of FastMixer's output for dumpsys Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); numCounterOffers = 0; index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); ALOG_ASSERT(index == 0); mTeeSink = teeSink; PipeReader *teeSource = new PipeReader(*teeSink); numCounterOffers = 0; index = teeSource->negotiate(offers, 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->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; state->mTeeSink = mTeeSink.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 } else { mFastMixer = NULL; } 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 != NULL) { FastMixerStateQueue *sq = mFastMixer->sq(); FastMixerState *state = sq->begin(); if (state->mCommand == FastMixerState::COLD_IDLE) { int32_t old = android_atomic_inc(&mFastMixerFutex); if (old == -1) { __futex_syscall3(&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*/); delete mFastMixer; #ifdef AUDIO_WATCHDOG if (mAudioWatchdog != 0) { mAudioWatchdog->requestExit(); mAudioWatchdog->requestExitAndWait(); mAudioWatchdog.clear(); } #endif } delete mAudioMixer; } 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) { #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 }; 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); } } } } bool AudioFlinger::PlaybackThread::threadLoop() { Vector< sp > tracksToRemove; standbyTime = systemTime(); #ifdef SRS_PROCESSING if (mType == MIXER) { POSTPRO_PATCH_ICS_OUTPROC_MIX_INIT(this, gettid()); } else if (mType == DUPLICATING) { POSTPRO_PATCH_ICS_OUTPROC_DUPE_INIT(this, gettid()); } #endif // MIXER nsecs_t lastWarning = 0; // DUPLICATING // FIXME could this be made local to while loop? writeFrames = 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(); while (!exitPending()) { cpuStats.sample(myName); Vector< sp > effectChains; processConfigEvents(); { // scope for mLock Mutex::Autolock _l(mLock); if (checkForNewParameters_l()) { cacheParameters_l(); } saveOutputTracks(); // put audio hardware into standby after short delay if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || isSuspended())) { if (!mStandby) { 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(); // 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; checkSilentMode_l(); standbyTime = systemTime() + standbyDelay; sleepTime = idleSleepTime; if (mType == MIXER) { sleepTimeShift = 0; } continue; } } // mMixerStatusIgnoringFastTracks is also updated internally mMixerStatus = prepareTracks_l(&tracksToRemove); // 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); } if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { threadLoop_mix(); } else { threadLoop_sleepTime(); } if (isSuspended()) { sleepTime = suspendSleepTimeUs(); mBytesWritten += mixBufferSize; } // only process effects if we're going to write if (sleepTime == 0) { for (size_t i = 0; i < effectChains.size(); i ++) { #ifdef QCOM_HARDWARE if (effectChains[i] != mAudioFlinger->mLPAEffectChain) #endif effectChains[i]->process_l(); } } // enable changes in effect chain unlockEffectChains(effectChains); // sleepTime == 0 means we must write to audio hardware if (sleepTime == 0) { #ifdef SRS_PROCESSING if (mType == MIXER) { POSTPRO_PATCH_ICS_OUTPROC_MIX_SAMPLES(this, mFormat, mMixBuffer, mixBufferSize, mSampleRate, mChannelCount); } else if (mType == DUPLICATING) { POSTPRO_PATCH_ICS_OUTPROC_DUPE_SAMPLES(this, mFormat, mMixBuffer, mixBufferSize, mSampleRate, mChannelCount); } #endif threadLoop_write(); if (mType == MIXER) { // write blocked detection nsecs_t now = systemTime(); nsecs_t delta = now - mLastWriteTime; if (!mStandby && delta > maxPeriod) { mNumDelayedWrites++; if ((now - lastWarning) > kWarningThrottleNs) { #if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) ScopedTrace st(ATRACE_TAG, "underrun"); #endif ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", ns2ms(delta), mNumDelayedWrites, this); lastWarning = now; } } } mStandby = false; } 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). } // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... if (mType == MIXER || mType == DIRECT) { // put output stream into standby mode if (!mStandby) { mOutput->stream->common.standby(&mOutput->stream->common); } } #ifdef SRS_PROCESSING if (mType == MIXER) { POSTPRO_PATCH_ICS_OUTPROC_MIX_EXIT(this, gettid()); } else if (mType == DUPLICATING) { POSTPRO_PATCH_ICS_OUTPROC_DUPE_EXIT(this, gettid()); } #endif releaseWakeLock(); ALOGV("Thread %p type %d exiting", this, mType); return false; } void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp >& tracksToRemove) { PlaybackThread::threadLoop_removeTracks(tracksToRemove); } void 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 != NULL) { 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) { __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); } #ifdef AUDIO_WATCHDOG if (mAudioWatchdog != 0) { mAudioWatchdog->resume(); } #endif } state->mCommand = FastMixerState::MIX_WRITE; sq->end(); sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); if (kUseFastMixer == FastMixer_Dynamic) { mNormalSink = mPipeSink; } } else { sq->end(false /*didModify*/); } } PlaybackThread::threadLoop_write(); } // shared by MIXER and DIRECT, overridden by DUPLICATING void AudioFlinger::PlaybackThread::threadLoop_write() { // FIXME rewrite to reduce number of system calls mLastWriteTime = systemTime(); mInWrite = true; int bytesWritten; // If an NBAIO sink is present, use it to write the normal mixer's submix if (mNormalSink != 0) { #define mBitShift 2 // FIXME size_t count = mixBufferSize >> mBitShift; #if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) Tracer::traceBegin(ATRACE_TAG, "write"); #endif // update the setpoint when gScreenState changes uint32_t screenState = gScreenState; 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(mMixBuffer, count); #if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) Tracer::traceEnd(ATRACE_TAG); #endif if (framesWritten > 0) { bytesWritten = framesWritten << mBitShift; } else { bytesWritten = framesWritten; } // otherwise use the HAL / AudioStreamOut directly } else { // Direct output thread. bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); } if (bytesWritten > 0) mBytesWritten += mixBufferSize; mNumWrites++; mInWrite = false; } void AudioFlinger::MixerThread::threadLoop_standby() { // Idle the fast mixer if it's currently running if (mFastMixer != NULL) { 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(); } // 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); } void AudioFlinger::MixerThread::threadLoop_mix() { // obtain the presentation timestamp of the next output buffer int64_t pts; status_t status = INVALID_OPERATION; #ifndef ICS_AUDIO_BLOB if (mNormalSink != 0) { status = mNormalSink->getNextWriteTimestamp(&pts); } else { status = mOutputSink->getNextWriteTimestamp(&pts); } #endif if (status != NO_ERROR) { pts = AudioBufferProvider::kInvalidPTS; } // mix buffers... mAudioMixer->process(pts); // 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)) { memset (mMixBuffer, 0, mixBufferSize); 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 > *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 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 != NULL) { sq = mFastMixer->sq(); state = sq->begin(); } for (size_t i=0 ; i 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())) { track->mUnderrunCount += recentUnderruns; } // 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->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_ON, &track->mCblk->flags); // remove from active list, but state remains ACTIVE [confusing but true] isActive = false; break; } // fall through case TrackBase::STOPPING_2: case TrackBase::PAUSED: case TrackBase::TERMINATED: 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 / audio_stream_frame_size(&mOutput->stream->common); 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_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->mSampleRate = track->mSampleRate; fastTrack->mChannelMask = track->mChannelMask; 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 = track->isMuted() ? 0 : 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_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 uint32_t minFrames = 1; if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { if (t->sampleRate() == (int)mSampleRate) { minFrames = mNormalFrameCount; } else { // +1 for rounding and +1 for additional sample needed for interpolation if(mSampleRate) minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; else { minFrames = 2; ALOGW("SampleRate is 0"); } // add frames already consumed but not yet released by the resampler // because cblk->framesReady() will include these frames minFrames += mAudioMixer->getUnreleasedFrames(track->name()); // 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(minFrames <= cblk->frameCount); } } if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && !track->isTerminated()) { //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); mixedTracks++; // track->mainBuffer() != mMixBuffer means there is an effect chain // connected to the track chain.clear(); if (track->mainBuffer() != mMixBuffer) { 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); } else if (cblk->server != 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, va; if (track->isMuted() || track->isPausing() || mStreamTypes[track->streamType()].mute) { vl = vr = va = 0; if (track->isPausing()) { track->setPaused(); } } else { // read original volumes with volume control float typeVolume = mStreamTypes[track->streamType()].volume; float v = masterVolume * typeVolume; uint32_t vlr = cblk->getVolumeLR(); vl = vlr & 0xFFFF; vr = vlr >> 16; // track volumes come from shared memory, so can't be trusted and must be clamped if (vl > MAX_GAIN_INT) { ALOGV("Track left volume out of range: %04X", vl); vl = MAX_GAIN_INT; } if (vr > MAX_GAIN_INT) { ALOGV("Track right volume out of range: %04X", vr); vr = MAX_GAIN_INT; } // now apply the master volume and stream type volume vl = (uint32_t)(v * vl) << 12; vr = (uint32_t)(v * vr) << 12; // assuming master volume and stream type volume each go up to 1.0, // vl and vr are now in 8.24 format uint16_t sendLevel = cblk->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; } va = (uint32_t)(v * sendLevel); } // 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; 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; } // Convert volumes from 8.24 to 4.12 format // This additional clamping is needed in case chain->setVolume_l() overshot vl = (vl + (1 << 11)) >> 12; if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; vr = (vr + (1 << 11)) >> 12; if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - // XXX: these things DON'T need to be done each time mAudioMixer->setBufferProvider(name, track); mAudioMixer->enable(name); mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); mAudioMixer->setParameter( name, AudioMixer::TRACK, AudioMixer::FORMAT, (void *)track->format()); mAudioMixer->setParameter( name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); mAudioMixer->setParameter( name, AudioMixer::RESAMPLE, AudioMixer::SAMPLE_RATE, (void *)(cblk->sampleRate)); 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 { // 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(); } //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, 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 / audio_stream_frame_size(&mOutput->stream->common); if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { if (track->isStopped()) { track->reset(); } tracksToRemove->add(track); } } else { track->mUnderrunCount++; // 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("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_ON, &cblk->flags); // 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; } sq->end(); } 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 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... count = tracksToRemove->size(); if (CC_UNLIKELY(count)) { for (size_t i=0 ; i& track = tracksToRemove->itemAt(i); mActiveTracks.remove(track); if (track->mainBuffer() != mMixBuffer) { 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); } } } // mix buffer must be cleared if all tracks are connected to an // effect chain as in this case the mixer will not write to // mix buffer and track effects will accumulate into it if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { // FIXME as a performance optimization, should remember previous zero status memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); } // if any fast tracks, then status is ready mMixerStatusIgnoringFastTracks = mixerStatus; if (fastTracks > 0) { mixerStatus = MIXER_TRACKS_READY; } return mixerStatus; } /* The derived values that are cached: - mixBufferSize 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() { mixBufferSize = mNormalFrameCount * mFrameSize; activeSleepTime = activeSleepTimeUs(); idleSleepTime = idleSleepTimeUs(); } void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) { ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", this, streamType, mTracks.size()); Mutex::Autolock _l(mLock); size_t size = mTracks.size(); for (size_t i = 0; i < size; i++) { sp t = mTracks[i]; if (t->streamType() == streamType) { android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); t->mCblk->cv.signal(); } } } // getTrackName_l() must be called with ThreadBase::mLock held int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) { return mAudioMixer->getTrackName(channelMask, 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); } // checkForNewParameters_l() must be called with ThreadBase::mLock held bool AudioFlinger::MixerThread::checkForNewParameters_l() { // if !&IDLE, holds the FastMixer state to restore after new parameters processed FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; bool reconfig = false; while (!mNewParameters.isEmpty()) { if (mFastMixer != NULL) { 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*/); } } status_t status = NO_ERROR; String8 keyValuePair = mNewParameters[0]; AudioParameter param = AudioParameter(keyValuePair); int value; #ifdef SRS_PROCESSING POSTPRO_PATCH_ICS_OUTPROC_MIX_ROUTE(this, param, value); #endif if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 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 { reconfig = true; } } if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { if (value != AUDIO_CHANNEL_OUT_STEREO) { status = BAD_VALUE; } else { 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. 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) { delete mAudioMixer; // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) mAudioMixer = NULL; readOutputParameters(); mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); for (size_t i = 0; i < mTracks.size() ; i++) { int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); if (name < 0) break; mTracks[i]->mName = name; // limit track sample rate to 2 x new output sample rate if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); } } sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); } } mNewParameters.removeAt(0); mParamStatus = status; mParamCond.signal(); // wait for condition with time out in case the thread calling ThreadBase::setParameters() // already timed out waiting for the status and will never signal the condition. mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); } if (!(previousCommand & FastMixerState::IDLE)) { ALOG_ASSERT(mFastMixer != NULL); 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& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; PlaybackThread::dumpInternals(fd, args); snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); result.append(buffer); write(fd, result.string(), result.size()); // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 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 // Write the tee output to a .wav file NBAIO_Source *teeSource = mTeeSource.get(); if (teeSource != NULL) { char teePath[64]; struct timeval tv; gettimeofday(&tv, NULL); struct tm tm; localtime_r(&tv.tv_sec, &tm); strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); if (teeFd >= 0) { char wavHeader[44]; memcpy(wavHeader, "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0", sizeof(wavHeader)); NBAIO_Format format = teeSource->format(); unsigned channelCount = Format_channelCount(format); ALOG_ASSERT(channelCount <= FCC_2); unsigned sampleRate = Format_sampleRate(format); wavHeader[22] = channelCount; // number of channels wavHeader[24] = sampleRate; // sample rate wavHeader[25] = sampleRate >> 8; wavHeader[32] = channelCount * 2; // block alignment write(teeFd, wavHeader, sizeof(wavHeader)); size_t total = 0; bool firstRead = true; for (;;) { #define TEE_SINK_READ 1024 short buffer[TEE_SINK_READ * FCC_2]; size_t count = TEE_SINK_READ; ssize_t actual = teeSource->read(buffer, count, AudioBufferProvider::kInvalidPTS); bool wasFirstRead = firstRead; firstRead = false; if (actual <= 0) { if (actual == (ssize_t) OVERRUN && wasFirstRead) { continue; } break; } ALOG_ASSERT(actual <= (ssize_t)count); write(teeFd, buffer, actual * channelCount * sizeof(short)); total += actual; } lseek(teeFd, (off_t) 4, SEEK_SET); uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; write(teeFd, &temp, sizeof(temp)); lseek(teeFd, (off_t) 40, SEEK_SET); temp = total * channelCount * sizeof(short); write(teeFd, &temp, sizeof(temp)); close(teeFd); fdprintf(fd, "FastMixer tee copied to %s\n", teePath); } else { fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); } } #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, AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) : PlaybackThread(audioFlinger, output, id, device, DIRECT) // mLeftVolFloat, mRightVolFloat { } AudioFlinger::DirectOutputThread::~DirectOutputThread() { } AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( Vector< sp > *tracksToRemove ) { sp trackToRemove; mixer_state mixerStatus = MIXER_IDLE; // find out which tracks need to be processed if (mActiveTracks.size() != 0) { sp t = mActiveTracks[0].promote(); // The track died recently if (t == 0) return MIXER_IDLE; Track* const track = t.get(); 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 uint32_t minFrames; if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { minFrames = mNormalFrameCount; } else { minFrames = 1; } if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && !track->isTerminated()) { //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); if (track->mFillingUpStatus == Track::FS_FILLED) { track->mFillingUpStatus = Track::FS_ACTIVE; mLeftVolFloat = mRightVolFloat = 0; if (track->mState == TrackBase::RESUMING) { track->mState = TrackBase::ACTIVE; } } // compute volume for this track float left, right; if (track->isMuted() || mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { left = right = 0; if (track->isPausing()) { track->setPaused(); } } else { float typeVolume = mStreamTypes[track->streamType()].volume; float v = mMasterVolume * typeVolume; uint32_t vlr = cblk->getVolumeLR(); float v_clamped = v * (vlr & 0xFFFF); if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; left = v_clamped/MAX_GAIN; v_clamped = v * (vlr >> 16); if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; right = v_clamped/MAX_GAIN; } 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()) { // Do not ramp volume if volume is controlled by effect mEffectChains[0]->setVolume_l(&vl, &vr); left = (float)vl / (1 << 24); right = (float)vr / (1 << 24); } mOutput->stream->set_volume(mOutput->stream, left, right); } // reset retry count track->mRetryCount = kMaxTrackRetriesDirect; mActiveTrack = t; mixerStatus = MIXER_TRACKS_READY; } else { // clear effect chain input buffer if an active track underruns to avoid sending // previous audio buffer again to effects if (!mEffectChains.isEmpty()) { mEffectChains[0]->clearInputBuffer(); } //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 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: implement behavior for compressed audio size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; size_t framesWritten = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { if (track->isStopped()) { track->reset(); } trackToRemove = 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("BUFFER TIMEOUT: remove(%d) from active list", track->name()); trackToRemove = track; } else { mixerStatus = MIXER_TRACKS_ENABLED; } } } } // FIXME merge this with similar code for removing multiple tracks // remove all the tracks that need to be... if (CC_UNLIKELY(trackToRemove != 0)) { tracksToRemove->add(trackToRemove); mActiveTracks.remove(trackToRemove); if (!mEffectChains.isEmpty()) { ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), trackToRemove->sessionId()); mEffectChains[0]->decActiveTrackCnt(); } if (trackToRemove->isTerminated()) { removeTrack_l(trackToRemove); } } return mixerStatus; } void AudioFlinger::DirectOutputThread::threadLoop_mix() { AudioBufferProvider::Buffer buffer; size_t frameCount = mFrameCount; int8_t *curBuf = (int8_t *)mMixBuffer; // output audio to hardware while (frameCount) { buffer.frameCount = frameCount; mActiveTrack->getNextBuffer(&buffer); if (CC_UNLIKELY(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); } 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)) { memset(mMixBuffer, 0, mFrameCount * mFrameSize); sleepTime = 0; } } // getTrackName_l() must be called with ThreadBase::mLock held int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) { return 0; } // deleteTrackName_l() must be called with ThreadBase::mLock held void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) { } // checkForNewParameters_l() must be called with ThreadBase::mLock held bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() { bool reconfig = false; while (!mNewParameters.isEmpty()) { status_t status = NO_ERROR; String8 keyValuePair = mNewParameters[0]; AudioParameter param = AudioParameter(keyValuePair); int value; 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(); sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); } } mNewParameters.removeAt(0); mParamStatus = status; mParamCond.signal(); // wait for condition with time out in case the thread calling ThreadBase::setParameters() // already timed out waiting for the status and will never signal the condition. mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); } 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 standbyDelay = microseconds(activeSleepTime*2); } // ---------------------------------------------------------------------------- AudioFlinger::DuplicatingThread::DuplicatingThread(const sp& 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 { memset(mMixBuffer, 0, mixBufferSize); } sleepTime = 0; writeFrames = mNormalFrameCount; 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; memset(mMixBuffer, 0, mixBufferSize); } else { // flush remaining overflow buffers in output tracks writeFrames = 0; } sleepTime = 0; } } void AudioFlinger::DuplicatingThread::threadLoop_write() { for (size_t i = 0; i < outputTracks.size(); i++) { outputTracks[i]->write(mMixBuffer, writeFrames); } mBytesWritten += mixBufferSize; } 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 int sampleRate = thread->sampleRate(); int frameCount = 0; if (sampleRate) frameCount = (3 * mNormalFrameCount * mSampleRate) / sampleRate; OutputTrack *outputTrack = new OutputTrack(thread, this, mSampleRate, mFormat, mChannelMask, frameCount); 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(); if (thread->getOutput() == mOutput) { mOutput = NULL; } 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 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 > &outputTracks) { for (size_t i = 0; i < outputTracks.size(); i++) { sp 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(); } // ---------------------------------------------------------------------------- // TrackBase constructor must be called with AudioFlinger::mLock held AudioFlinger::ThreadBase::TrackBase::TrackBase( ThreadBase *thread, const sp& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, #ifdef QCOM_ENHANCED_AUDIO uint32_t flags, #endif const sp& sharedBuffer, int sessionId) : RefBase(), mThread(thread), mClient(client), mCblk(NULL), // mBuffer // mBufferEnd mFrameCount(0), mState(IDLE), mSampleRate(sampleRate), mFormat(format), mStepServerFailed(false), #ifdef QCOM_ENHANCED_AUDIO mFlags(0), #endif mSessionId(sessionId) // mChannelCount // mChannelMask { ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); size_t size = sizeof(audio_track_cblk_t); uint8_t channelCount = popcount(channelMask); #ifdef QCOM_ENHANCED_AUDIO size_t bufferSize = 0; if ((int16_t)flags == 0x1) { bufferSize = frameCount*channelCount*sizeof(int16_t); } else { if ( (format == AUDIO_FORMAT_PCM_16_BIT) || (format == AUDIO_FORMAT_PCM_8_BIT)) { bufferSize = frameCount*channelCount*sizeof(int16_t); } else if (format == AUDIO_FORMAT_AMR_NB) { bufferSize = frameCount*channelCount*32; // full rate frame size } else if (format == AUDIO_FORMAT_EVRC) { bufferSize = frameCount*channelCount*23; // full rate frame size } else if (format == AUDIO_FORMAT_QCELP) { bufferSize = frameCount*channelCount*35; // full rate frame size } else if (format == AUDIO_FORMAT_AAC) { bufferSize = frameCount*2048; // full rate frame size } else if (format == AUDIO_FORMAT_AMR_WB) { bufferSize = frameCount*channelCount*61; // full rate frame size } } #else size_t bufferSize = frameCount*channelCount*sizeof(int16_t); #endif if (sharedBuffer == 0) { size += bufferSize; } if (client != NULL) { mCblkMemory = client->heap()->allocate(size); if (mCblkMemory != 0) { mCblk = static_cast(mCblkMemory->pointer()); if (mCblk != NULL) { // construct the shared structure in-place. new(mCblk) audio_track_cblk_t(); // clear all buffers mCblk->frameCount = frameCount; mCblk->sampleRate = sampleRate; // uncomment the following lines to quickly test 32-bit wraparound // mCblk->user = 0xffff0000; // mCblk->server = 0xffff0000; // mCblk->userBase = 0xffff0000; // mCblk->serverBase = 0xffff0000; mChannelCount = channelCount; mChannelMask = channelMask; if (sharedBuffer == 0) { mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); #ifdef QCOM_ENHANCED_AUDIO if ((int16_t)flags == 0x1) { bufferSize = frameCount*channelCount*sizeof(int16_t); } else { if ((format == AUDIO_FORMAT_PCM_16_BIT) || (format == AUDIO_FORMAT_PCM_8_BIT)) { memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); } else if (format == AUDIO_FORMAT_AMR_NB) { memset(mBuffer, 0, frameCount*channelCount*32); // full rate frame size } else if (format == AUDIO_FORMAT_EVRC) { memset(mBuffer, 0, frameCount*channelCount*23); // full rate frame size } else if (format == AUDIO_FORMAT_QCELP) { memset(mBuffer, 0, frameCount*channelCount*35); // full rate frame size } else if (format == AUDIO_FORMAT_AAC) { memset(mBuffer, 0, frameCount*2048); // full rate frame size } else if (format == AUDIO_FORMAT_AMR_WB) { memset(mBuffer, 0, frameCount*channelCount*61); // full rate frame size } } #else memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); #endif // Force underrun condition to avoid false underrun callback until first data is // written to buffer (other flags are cleared) mCblk->flags = CBLK_UNDERRUN_ON; } else { mBuffer = sharedBuffer->pointer(); } mBufferEnd = (uint8_t *)mBuffer + bufferSize; } } else { ALOGE("not enough memory for AudioTrack size=%u", size); client->heap()->dump("AudioTrack"); return; } } else { mCblk = (audio_track_cblk_t *)(new uint8_t[size]); // construct the shared structure in-place. new(mCblk) audio_track_cblk_t(); // clear all buffers mCblk->frameCount = frameCount; mCblk->sampleRate = sampleRate; // uncomment the following lines to quickly test 32-bit wraparound // mCblk->user = 0xffff0000; // mCblk->server = 0xffff0000; // mCblk->userBase = 0xffff0000; // mCblk->serverBase = 0xffff0000; mChannelCount = channelCount; mChannelMask = channelMask; mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); // Force underrun condition to avoid false underrun callback until first data is // written to buffer (other flags are cleared) mCblk->flags = CBLK_UNDERRUN_ON; mBufferEnd = (uint8_t *)mBuffer + bufferSize; } } AudioFlinger::ThreadBase::TrackBase::~TrackBase() { if (mCblk != NULL) { if (mClient == 0) { delete mCblk; } else { mCblk->~audio_track_cblk_t(); // destroy our shared-structure. } } mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to if (mClient != 0) { // Client destructor must run with AudioFlinger mutex locked Mutex::Autolock _l(mClient->audioFlinger()->mLock); // If the client's reference count drops to zero, the associated destructor // must run with AudioFlinger lock held. Thus the explicit clear() rather than // relying on the automatic clear() at end of scope. mClient.clear(); } } // AudioBufferProvider interface // getNextBuffer() = 0; // This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) { buffer->raw = NULL; mFrameCount = buffer->frameCount; // FIXME See note at getNextBuffer() (void) step(); // ignore return value of step() buffer->frameCount = 0; } bool AudioFlinger::ThreadBase::TrackBase::step() { bool result; audio_track_cblk_t* cblk = this->cblk(); result = cblk->stepServer(mFrameCount); if (!result) { ALOGV("stepServer failed acquiring cblk mutex"); mStepServerFailed = true; } return result; } void AudioFlinger::ThreadBase::TrackBase::reset() { audio_track_cblk_t* cblk = this->cblk(); cblk->user = 0; cblk->server = 0; cblk->userBase = 0; cblk->serverBase = 0; mStepServerFailed = false; ALOGV("TrackBase::reset"); } int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { return (int)mCblk->sampleRate; } void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { audio_track_cblk_t* cblk = this->cblk(); size_t frameSize = cblk->frameSize; int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; int8_t *bufferEnd = bufferStart + frames * frameSize; // Check validity of returned pointer in case the track control block would have been corrupted. #ifdef QCOM_ENHANCED_AUDIO if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd){ ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ server %u, serverBase %u, user %u, userBase %u", bufferStart, bufferEnd, mBuffer, mBufferEnd, cblk->server, cblk->serverBase, cblk->user, cblk->userBase); return 0; } #else ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), "TrackBase::getBuffer buffer out of range:\n" " start: %p, end %p , mBuffer %p mBufferEnd %p\n" " server %u, serverBase %u, user %u, userBase %u, frameSize %d", cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); #endif return bufferStart; } status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp& event) { mSyncEvents.add(event); return NO_ERROR; } // ---------------------------------------------------------------------------- // Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held AudioFlinger::PlaybackThread::Track::Track( PlaybackThread *thread, const sp& client, audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, const sp& sharedBuffer, int sessionId, IAudioFlinger::track_flags_t flags) : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, #ifdef QCOM_ENHANCED_AUDIO ((audio_stream_type_t)streamType == AUDIO_STREAM_VOICE_CALL)?0x1:0x0, #endif sharedBuffer, sessionId), mMute(false), mFillingUpStatus(FS_INVALID), // mRetryCount initialized later when needed mSharedBuffer(sharedBuffer), mStreamType(streamType), mName(-1), // see note below mMainBuffer(thread->mixBuffer()), mAuxBuffer(NULL), mAuxEffectId(0), mHasVolumeController(false), mPresentationCompleteFrames(0), mFlags(flags), mFastIndex(-1), mUnderrunCount(0), mCachedVolume(1.0) { if (mCblk != NULL) { // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack #ifdef QCOM_ENHANCED_AUDIO if ((audio_stream_type_t)streamType == AUDIO_STREAM_VOICE_CALL) mCblk->frameSize = mChannelCount * sizeof(int16_t); else #endif mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); // to avoid leaking a track name, do not allocate one unless there is an mCblk mName = thread->getTrackName_l(channelMask, sessionId); mCblk->mName = mName; if (mName < 0) { ALOGE("no more track names available"); return; } // only allocate a fast track index if we were able to allocate a normal track name if (flags & IAudioFlinger::TRACK_FAST) { mCblk->flags |= CBLK_FAST; // atomic op not needed yet ALOG_ASSERT(thread->mFastTrackAvailMask != 0); int i = __builtin_ctz(thread->mFastTrackAvailMask); ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); // FIXME This is too eager. We allocate a fast track index before the // fast track becomes active. Since fast tracks are a scarce resource, // this means we are potentially denying other more important fast tracks from // being created. It would be better to allocate the index dynamically. mFastIndex = i; mCblk->mName = i; // Read the initial underruns because this field is never cleared by the fast mixer mObservedUnderruns = thread->getFastTrackUnderruns(i); thread->mFastTrackAvailMask &= ~(1 << i); } } ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); } AudioFlinger::PlaybackThread::Track::~Track() { ALOGV("PlaybackThread::Track destructor"); } void AudioFlinger::PlaybackThread::Track::destroy() { // NOTE: destroyTrack_l() can remove a strong reference to this Track // by removing it from mTracks vector, so there is a risk that this Tracks's // destructor is called. As the destructor needs to lock mLock, // we must acquire a strong reference on this Track before locking mLock // here so that the destructor is called only when exiting this function. // On the other hand, as long as Track::destroy() is only called by // TrackHandle destructor, the TrackHandle still holds a strong ref on // this Track with its member mTrack. sp keep(this); { // scope for mLock sp thread = mThread.promote(); if (thread != 0) { if (!isOutputTrack()) { if (mState == ACTIVE || mState == RESUMING) { AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); #ifdef ADD_BATTERY_DATA // to track the speaker usage addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); #endif } AudioSystem::releaseOutput(thread->id()); } Mutex::Autolock _l(thread->mLock); PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); playbackThread->destroyTrack_l(this); } } } /*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) { result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " " Server User Main buf Aux Buf Flags Underruns\n"); } void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) { uint32_t vlr = mCblk->getVolumeLR(); if (isFastTrack()) { sprintf(buffer, " F %2d", mFastIndex); } else { sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); } track_state state = mState; char stateChar; switch (state) { case IDLE: stateChar = 'I'; break; case TERMINATED: stateChar = 'T'; break; case STOPPING_1: stateChar = 's'; break; case STOPPING_2: stateChar = '5'; break; case STOPPED: stateChar = 'S'; break; case RESUMING: stateChar = 'R'; break; case ACTIVE: stateChar = 'A'; break; case PAUSING: stateChar = 'p'; break; case PAUSED: stateChar = 'P'; break; case FLUSHED: stateChar = 'F'; break; default: stateChar = '?'; break; } char nowInUnderrun; switch (mObservedUnderruns.mBitFields.mMostRecent) { case UNDERRUN_FULL: nowInUnderrun = ' '; break; case UNDERRUN_PARTIAL: nowInUnderrun = '<'; break; case UNDERRUN_EMPTY: nowInUnderrun = '*'; break; default: nowInUnderrun = '?'; break; } snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", (mClient == 0) ? getpid_cached : mClient->pid(), mStreamType, mFormat, mChannelMask, mSessionId, mFrameCount, mCblk->frameCount, stateChar, mMute, mFillingUpStatus, mCblk->sampleRate, 20.0 * log10((vlr & 0xFFFF) / 4096.0), 20.0 * log10((vlr >> 16) / 4096.0), mCblk->server, mCblk->user, (int)mMainBuffer, (int)mAuxBuffer, mCblk->flags, mUnderrunCount, nowInUnderrun); } // AudioBufferProvider interface status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( AudioBufferProvider::Buffer* buffer, int64_t pts) { audio_track_cblk_t* cblk = this->cblk(); uint32_t framesReady; uint32_t framesReq = buffer->frameCount; // Check if last stepServer failed, try to step now if (mStepServerFailed) { // FIXME When called by fast mixer, this takes a mutex with tryLock(). // Since the fast mixer is higher priority than client callback thread, // it does not result in priority inversion for client. // But a non-blocking solution would be preferable to avoid // fast mixer being unable to tryLock(), and // to avoid the extra context switches if the client wakes up, // discovers the mutex is locked, then has to wait for fast mixer to unlock. if (!step()) goto getNextBuffer_exit; ALOGV("stepServer recovered"); mStepServerFailed = false; } // FIXME Same as above framesReady = cblk->framesReady(); if (CC_LIKELY(framesReady)) { uint32_t s = cblk->server; uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; if (framesReq > framesReady) { framesReq = framesReady; } if (framesReq > bufferEnd - s) { framesReq = bufferEnd - s; } buffer->raw = getBuffer(s, framesReq); buffer->frameCount = framesReq; return NO_ERROR; } getNextBuffer_exit: buffer->raw = NULL; buffer->frameCount = 0; ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); return NOT_ENOUGH_DATA; } // Note that framesReady() takes a mutex on the control block using tryLock(). // This could result in priority inversion if framesReady() is called by the normal mixer, // as the normal mixer thread runs at lower // priority than the client's callback thread: there is a short window within framesReady() // during which the normal mixer could be preempted, and the client callback would block. // Another problem can occur if framesReady() is called by the fast mixer: // the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. // FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. size_t AudioFlinger::PlaybackThread::Track::framesReady() const { return mCblk->framesReady(); } // Don't call for fast tracks; the framesReady() could result in priority inversion bool AudioFlinger::PlaybackThread::Track::isReady() const { if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; if (framesReady() >= mCblk->frameCount || (mCblk->flags & CBLK_FORCEREADY_MSK)) { mFillingUpStatus = FS_FILLED; android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); return true; } return false; } status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, int triggerSession) { status_t status = NO_ERROR; ALOGV("start(%d), calling pid %d session %d", mName, IPCThreadState::self()->getCallingPid(), mSessionId); sp thread = mThread.promote(); if (thread != 0) { Mutex::Autolock _l(thread->mLock); track_state state = mState; // here the track could be either new, or restarted // in both cases "unstop" the track if (mState == PAUSED) { mState = TrackBase::RESUMING; ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); } else { mState = TrackBase::ACTIVE; ALOGV("? => ACTIVE (%d) on thread %p", mName, this); } if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { thread->mLock.unlock(); status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); thread->mLock.lock(); #ifdef ADD_BATTERY_DATA // to track the speaker usage if (status == NO_ERROR) { addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); } #endif } if (status == NO_ERROR) { PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); playbackThread->addTrack_l(this); } else { mState = state; triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); } } else { status = BAD_VALUE; } return status; } void AudioFlinger::PlaybackThread::Track::stop() { ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); sp thread = mThread.promote(); if (thread != 0) { Mutex::Autolock _l(thread->mLock); track_state state = mState; if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { // If the track is not active (PAUSED and buffers full), flush buffers PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); if (playbackThread->mActiveTracks.indexOf(this) < 0) { reset(); mState = STOPPED; } else if (!isFastTrack()) { mState = STOPPED; } else { // prepareTracks_l() will set state to STOPPING_2 after next underrun, // and then to STOPPED and reset() when presentation is complete mState = STOPPING_1; } ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); } if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { thread->mLock.unlock(); AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); thread->mLock.lock(); #ifdef ADD_BATTERY_DATA // to track the speaker usage addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); #endif } } } void AudioFlinger::PlaybackThread::Track::pause() { ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); sp thread = mThread.promote(); if (thread != 0) { Mutex::Autolock _l(thread->mLock); if (mState == ACTIVE || mState == RESUMING) { mState = PAUSING; ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); if (!isOutputTrack()) { thread->mLock.unlock(); AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); thread->mLock.lock(); #ifdef ADD_BATTERY_DATA // to track the speaker usage addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); #endif } } } } void AudioFlinger::PlaybackThread::Track::flush() { ALOGV("flush(%d)", mName); sp thread = mThread.promote(); if (thread != 0) { Mutex::Autolock _l(thread->mLock); if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { return; } // No point remaining in PAUSED state after a flush => go to // FLUSHED state mState = FLUSHED; // do not reset the track if it is still in the process of being stopped or paused. // this will be done by prepareTracks_l() when the track is stopped. // prepareTracks_l() will see mState == FLUSHED, then // remove from active track list, reset(), and trigger presentation complete PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); if (playbackThread->mActiveTracks.indexOf(this) < 0) { reset(); } } } void AudioFlinger::PlaybackThread::Track::reset() { // Do not reset twice to avoid discarding data written just after a flush and before // the audioflinger thread detects the track is stopped. if (!mResetDone) { TrackBase::reset(); // Force underrun condition to avoid false underrun callback until first data is // written to buffer android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); mFillingUpStatus = FS_FILLING; mResetDone = true; if (mState == FLUSHED) { mState = IDLE; } } } void AudioFlinger::PlaybackThread::Track::mute(bool muted) { mMute = muted; } status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) { status_t status = DEAD_OBJECT; sp thread = mThread.promote(); if (thread != 0) { PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); sp af = mClient->audioFlinger(); Mutex::Autolock _l(af->mLock); sp srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { Mutex::Autolock _dl(playbackThread->mLock); Mutex::Autolock _sl(srcThread->mLock); sp chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); if (chain == 0) { return INVALID_OPERATION; } sp effect = chain->getEffectFromId_l(EffectId); if (effect == 0) { return INVALID_OPERATION; } srcThread->removeEffect_l(effect); playbackThread->addEffect_l(effect); // removeEffect_l() has stopped the effect if it was active so it must be restarted if (effect->state() == EffectModule::ACTIVE || effect->state() == EffectModule::STOPPING) { effect->start(); } sp dstChain = effect->chain().promote(); if (dstChain == 0) { srcThread->addEffect_l(effect); return INVALID_OPERATION; } AudioSystem::unregisterEffect(effect->id()); AudioSystem::registerEffect(&effect->desc(), srcThread->id(), dstChain->strategy(), AUDIO_SESSION_OUTPUT_MIX, effect->id()); } status = playbackThread->attachAuxEffect(this, EffectId); } return status; } void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) { mAuxEffectId = EffectId; mAuxBuffer = buffer; } bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, size_t audioHalFrames) { // a track is considered presented when the total number of frames written to audio HAL // corresponds to the number of frames written when presentationComplete() is called for the // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. if (mPresentationCompleteFrames == 0) { mPresentationCompleteFrames = framesWritten + audioHalFrames; ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", mPresentationCompleteFrames, audioHalFrames); } if (framesWritten >= mPresentationCompleteFrames) { ALOGV("presentationComplete() session %d complete: framesWritten %d", mSessionId, framesWritten); triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); return true; } return false; } void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) { for (int i = 0; i < (int)mSyncEvents.size(); i++) { if (mSyncEvents[i]->type() == type) { mSyncEvents[i]->trigger(); mSyncEvents.removeAt(i); i--; } } } // implement VolumeBufferProvider interface uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() { // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); uint32_t vlr = mCblk->getVolumeLR(); uint32_t vl = vlr & 0xFFFF; uint32_t vr = vlr >> 16; // track volumes come from shared memory, so can't be trusted and must be clamped if (vl > MAX_GAIN_INT) { vl = MAX_GAIN_INT; } if (vr > MAX_GAIN_INT) { vr = MAX_GAIN_INT; } // now apply the cached master volume and stream type volume; // this is trusted but lacks any synchronization or barrier so may be stale float v = mCachedVolume; vl *= v; vr *= v; // re-combine into U4.16 vlr = (vr << 16) | (vl & 0xFFFF); // FIXME look at mute, pause, and stop flags return vlr; } status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp& event) { if (mState == TERMINATED || mState == PAUSED || ((framesReady() == 0) && ((mSharedBuffer != 0) || (mState == STOPPED)))) { ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); event->cancel(); return INVALID_OPERATION; } (void) TrackBase::setSyncEvent(event); return NO_ERROR; } // timed audio tracks sp AudioFlinger::PlaybackThread::TimedTrack::create( PlaybackThread *thread, const sp& client, audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, const sp& sharedBuffer, int sessionId) { if (!client->reserveTimedTrack()) return 0; return new TimedTrack( thread, client, streamType, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId); } AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( PlaybackThread *thread, const sp& client, audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, const sp& sharedBuffer, int sessionId) : Track(thread, client, streamType, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), mQueueHeadInFlight(false), mTrimQueueHeadOnRelease(false), mFramesPendingInQueue(0), mTimedSilenceBuffer(NULL), mTimedSilenceBufferSize(0), mTimedAudioOutputOnTime(false), mMediaTimeTransformValid(false) { LocalClock lc; mLocalTimeFreq = lc.getLocalFreq(); mLocalTimeToSampleTransform.a_zero = 0; mLocalTimeToSampleTransform.b_zero = 0; mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, &mLocalTimeToSampleTransform.a_to_b_denom); mMediaTimeToSampleTransform.a_zero = 0; mMediaTimeToSampleTransform.b_zero = 0; mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; mMediaTimeToSampleTransform.a_to_b_denom = 1000000; LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, &mMediaTimeToSampleTransform.a_to_b_denom); } AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { mClient->releaseTimedTrack(); delete [] mTimedSilenceBuffer; } status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( size_t size, sp* buffer) { Mutex::Autolock _l(mTimedBufferQueueLock); trimTimedBufferQueue_l(); // lazily initialize the shared memory heap for timed buffers if (mTimedMemoryDealer == NULL) { const int kTimedBufferHeapSize = 512 << 10; mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, "AudioFlingerTimed"); if (mTimedMemoryDealer == NULL) return NO_MEMORY; } sp newBuffer = mTimedMemoryDealer->allocate(size); if (newBuffer == NULL) { newBuffer = mTimedMemoryDealer->allocate(size); if (newBuffer == NULL) return NO_MEMORY; } *buffer = newBuffer; return NO_ERROR; } // caller must hold mTimedBufferQueueLock void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { int64_t mediaTimeNow; { Mutex::Autolock mttLock(mMediaTimeTransformLock); if (!mMediaTimeTransformValid) return; int64_t targetTimeNow; status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) ? mCCHelper.getCommonTime(&targetTimeNow) : mCCHelper.getLocalTime(&targetTimeNow); if (OK != res) return; if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, &mediaTimeNow)) { return; } } size_t trimEnd; for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { int64_t bufEnd; if ((trimEnd + 1) < mTimedBufferQueue.size()) { // We have a next buffer. Just use its PTS as the PTS of the frame // following the last frame in this buffer. If the stream is sparse // (ie, there are deliberate gaps left in the stream which should be // filled with silence by the TimedAudioTrack), then this can result // in one extra buffer being left un-trimmed when it could have // been. In general, this is not typical, and we would rather // optimized away the TS calculation below for the more common case // where PTSes are contiguous. bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); } else { // We have no next buffer. Compute the PTS of the frame following // the last frame in this buffer by computing the duration of of // this frame in media time units and adding it to the PTS of the // buffer. int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() / mCblk->frameSize; if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, &bufEnd)) { ALOGE("Failed to convert frame count of %lld to media time" " duration" " (scale factor %d/%u) in %s", frameCount, mMediaTimeToSampleTransform.a_to_b_numer, mMediaTimeToSampleTransform.a_to_b_denom, __PRETTY_FUNCTION__); break; } bufEnd += mTimedBufferQueue[trimEnd].pts(); } if (bufEnd > mediaTimeNow) break; // Is the buffer we want to use in the middle of a mix operation right // now? If so, don't actually trim it. Just wait for the releaseBuffer // from the mixer which should be coming back shortly. if (!trimEnd && mQueueHeadInFlight) { mTrimQueueHeadOnRelease = true; } } size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; if (trimStart < trimEnd) { // Update the bookkeeping for framesReady() for (size_t i = trimStart; i < trimEnd; ++i) { updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); } // Now actually remove the buffers from the queue. mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); } } void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( const char* logTag) { ALOG_ASSERT(mTimedBufferQueue.size() > 0, "%s called (reason \"%s\"), but timed buffer queue has no" " elements to trim.", __FUNCTION__, logTag); updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); mTimedBufferQueue.removeAt(0); } void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( const TimedBuffer& buf, const char* logTag) { uint32_t bufBytes = buf.buffer()->size(); uint32_t consumedAlready = buf.position(); ALOG_ASSERT(consumedAlready <= bufBytes, "Bad bookkeeping while updating frames pending. Timed buffer is" " only %u bytes long, but claims to have consumed %u" " bytes. (update reason: \"%s\")", bufBytes, consumedAlready, logTag); uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, "Bad bookkeeping while updating frames pending. Should have at" " least %u queued frames, but we think we have only %u. (update" " reason: \"%s\")", bufFrames, mFramesPendingInQueue, logTag); mFramesPendingInQueue -= bufFrames; } status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( const sp& buffer, int64_t pts) { { Mutex::Autolock mttLock(mMediaTimeTransformLock); if (!mMediaTimeTransformValid) return INVALID_OPERATION; } Mutex::Autolock _l(mTimedBufferQueueLock); uint32_t bufFrames = buffer->size() / mCblk->frameSize; mFramesPendingInQueue += bufFrames; mTimedBufferQueue.add(TimedBuffer(buffer, pts)); return NO_ERROR; } status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, target); if (!(target == TimedAudioTrack::LOCAL_TIME || target == TimedAudioTrack::COMMON_TIME)) { return BAD_VALUE; } Mutex::Autolock lock(mMediaTimeTransformLock); mMediaTimeTransform = xform; mMediaTimeTransformTarget = target; mMediaTimeTransformValid = true; return NO_ERROR; } #define min(a, b) ((a) < (b) ? (a) : (b)) // implementation of getNextBuffer for tracks whose buffers have timestamps status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( AudioBufferProvider::Buffer* buffer, int64_t pts) { if (pts == AudioBufferProvider::kInvalidPTS) { buffer->raw = NULL; buffer->frameCount = 0; mTimedAudioOutputOnTime = false; return INVALID_OPERATION; } Mutex::Autolock _l(mTimedBufferQueueLock); ALOG_ASSERT(!mQueueHeadInFlight, "getNextBuffer called without releaseBuffer!"); while (true) { // if we have no timed buffers, then fail if (mTimedBufferQueue.isEmpty()) { buffer->raw = NULL; buffer->frameCount = 0; return NOT_ENOUGH_DATA; } TimedBuffer& head = mTimedBufferQueue.editItemAt(0); // calculate the PTS of the head of the timed buffer queue expressed in // local time int64_t headLocalPTS; { Mutex::Autolock mttLock(mMediaTimeTransformLock); ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); if (mMediaTimeTransform.a_to_b_denom == 0) { // the transform represents a pause, so yield silence timedYieldSilence_l(buffer->frameCount, buffer); return NO_ERROR; } int64_t transformedPTS; if (!mMediaTimeTransform.doForwardTransform(head.pts(), &transformedPTS)) { // the transform failed. this shouldn't happen, but if it does // then just drop this buffer ALOGW("timedGetNextBuffer transform failed"); buffer->raw = NULL; buffer->frameCount = 0; trimTimedBufferQueueHead_l("getNextBuffer; no transform"); return NO_ERROR; } if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, &headLocalPTS)) { buffer->raw = NULL; buffer->frameCount = 0; return INVALID_OPERATION; } } else { headLocalPTS = transformedPTS; } } // adjust the head buffer's PTS to reflect the portion of the head buffer // that has already been consumed int64_t effectivePTS = headLocalPTS + ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); // Calculate the delta in samples between the head of the input buffer // queue and the start of the next output buffer that will be written. // If the transformation fails because of over or underflow, it means // that the sample's position in the output stream is so far out of // whack that it should just be dropped. int64_t sampleDelta; if (llabs(effectivePTS - pts) >= (static_cast(1) << 31)) { ALOGV("*** head buffer is too far from PTS: dropped buffer"); trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" " mix"); continue; } if (!mLocalTimeToSampleTransform.doForwardTransform( (effectivePTS - pts) << 32, &sampleDelta)) { ALOGV("*** too late during sample rate transform: dropped buffer"); trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); continue; } ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" " sampleDelta=[%d.%08x]", head.pts(), head.position(), pts, static_cast((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), static_cast(sampleDelta & 0xFFFFFFFF)); // if the delta between the ideal placement for the next input sample and // the current output position is within this threshold, then we will // concatenate the next input samples to the previous output const int64_t kSampleContinuityThreshold = (static_cast(sampleRate()) << 32) / 250; // if this is the first buffer of audio that we're emitting from this track // then it should be almost exactly on time. const int64_t kSampleStartupThreshold = 1LL << 32; if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { // the next input is close enough to being on time, so concatenate it // with the last output timedYieldSamples_l(buffer); ALOGVV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); return NO_ERROR; } // Looks like our output is not on time. Reset our on timed status. // Next time we mix samples from our input queue, then should be within // the StartupThreshold. mTimedAudioOutputOnTime = false; if (sampleDelta > 0) { // the gap between the current output position and the proper start of // the next input sample is too big, so fill it with silence uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; timedYieldSilence_l(framesUntilNextInput, buffer); ALOGV("*** silence: frameCount=%u", buffer->frameCount); return NO_ERROR; } else { // the next input sample is late uint32_t lateFrames = static_cast(-((sampleDelta + 0x80000000) >> 32)); size_t onTimeSamplePosition = head.position() + lateFrames * mCblk->frameSize; if (onTimeSamplePosition > head.buffer()->size()) { // all the remaining samples in the head are too late, so // drop it and move on ALOGV("*** too late: dropped buffer"); trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); continue; } else { // skip over the late samples head.setPosition(onTimeSamplePosition); // yield the available samples timedYieldSamples_l(buffer); ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); return NO_ERROR; } } } } // Yield samples from the timed buffer queue head up to the given output // buffer's capacity. // // Caller must hold mTimedBufferQueueLock void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( AudioBufferProvider::Buffer* buffer) { const TimedBuffer& head = mTimedBufferQueue[0]; buffer->raw = (static_cast(head.buffer()->pointer()) + head.position()); uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / mCblk->frameSize); size_t framesRequested = buffer->frameCount; buffer->frameCount = min(framesLeftInHead, framesRequested); mQueueHeadInFlight = true; mTimedAudioOutputOnTime = true; } // Yield samples of silence up to the given output buffer's capacity // // Caller must hold mTimedBufferQueueLock void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { // lazily allocate a buffer filled with silence if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { delete [] mTimedSilenceBuffer; mTimedSilenceBufferSize = numFrames * mCblk->frameSize; mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); } buffer->raw = mTimedSilenceBuffer; size_t framesRequested = buffer->frameCount; buffer->frameCount = min(numFrames, framesRequested); mTimedAudioOutputOnTime = false; } // AudioBufferProvider interface void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( AudioBufferProvider::Buffer* buffer) { Mutex::Autolock _l(mTimedBufferQueueLock); // If the buffer which was just released is part of the buffer at the head // of the queue, be sure to update the amt of the buffer which has been // consumed. If the buffer being returned is not part of the head of the // queue, its either because the buffer is part of the silence buffer, or // because the head of the timed queue was trimmed after the mixer called // getNextBuffer but before the mixer called releaseBuffer. if (buffer->raw == mTimedSilenceBuffer) { ALOG_ASSERT(!mQueueHeadInFlight, "Queue head in flight during release of silence buffer!"); goto done; } ALOG_ASSERT(mQueueHeadInFlight, "TimedTrack::releaseBuffer of non-silence buffer, but no queue" " head in flight."); if (mTimedBufferQueue.size()) { TimedBuffer& head = mTimedBufferQueue.editItemAt(0); void* start = head.buffer()->pointer(); void* end = reinterpret_cast( reinterpret_cast(head.buffer()->pointer()) + head.buffer()->size()); ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), "released buffer not within the head of the timed buffer" " queue; qHead = [%p, %p], released buffer = %p", start, end, buffer->raw); head.setPosition(head.position() + (buffer->frameCount * mCblk->frameSize)); mQueueHeadInFlight = false; ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, "Bad bookkeeping during releaseBuffer! Should have at" " least %u queued frames, but we think we have only %u", buffer->frameCount, mFramesPendingInQueue); mFramesPendingInQueue -= buffer->frameCount; if ((static_cast(head.position()) >= head.buffer()->size()) || mTrimQueueHeadOnRelease) { trimTimedBufferQueueHead_l("releaseBuffer"); mTrimQueueHeadOnRelease = false; } } else { LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" " buffers in the timed buffer queue"); } done: buffer->raw = 0; buffer->frameCount = 0; } size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { Mutex::Autolock _l(mTimedBufferQueueLock); return mFramesPendingInQueue; } AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() : mPTS(0), mPosition(0) {} AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( const sp& buffer, int64_t pts) : mBuffer(buffer), mPTS(pts), mPosition(0) {} // ---------------------------------------------------------------------------- // RecordTrack constructor must be called with AudioFlinger::mLock held AudioFlinger::RecordThread::RecordTrack::RecordTrack( RecordThread *thread, const sp& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, #ifdef QCOM_ENHANCED_AUDIO uint32_t flags, #endif int sessionId) : TrackBase(thread, client, sampleRate, format,channelMask,frameCount, #ifdef QCOM_ENHANCED_AUDIO ((audio_source_t)((int16_t)flags) == AUDIO_SOURCE_VOICE_COMMUNICATION) ? ((flags & 0xffff0000)| 0x1) : ((flags & 0xffff0000)), #endif 0 /*sharedBuffer*/, sessionId), mOverflow(false) { uint8_t channelCount = popcount(channelMask); if (mCblk != NULL) { #ifdef QCOM_ENHANCED_AUDIO ALOGV("RecordTrack constructor, size %d flags %d", (int)mBufferEnd - (int)mBuffer,flags); if ((audio_source_t)((int16_t)flags) == AUDIO_SOURCE_VOICE_COMMUNICATION) { mCblk->frameSize = mChannelCount * sizeof(int16_t); } else { if (format == AUDIO_FORMAT_AMR_NB) { mCblk->frameSize = channelCount * 32; } else if (format == AUDIO_FORMAT_EVRC) { mCblk->frameSize = channelCount * 23; } else if (format == AUDIO_FORMAT_QCELP) { mCblk->frameSize = channelCount * 35; } else if (format == AUDIO_FORMAT_AAC) { mCblk->frameSize = 2048; } else if (format == AUDIO_FORMAT_PCM_16_BIT) { mCblk->frameSize = mChannelCount * sizeof(int16_t); } else if (format == AUDIO_FORMAT_PCM_8_BIT) { mCblk->frameSize = mChannelCount * sizeof(int8_t); } else if (format == AUDIO_FORMAT_AMR_WB) { mCblk->frameSize = channelCount * 61; } else { mCblk->frameSize = sizeof(int8_t); } } #else ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); if (format == AUDIO_FORMAT_PCM_16_BIT) { mCblk->frameSize = mChannelCount * sizeof(int16_t); } else if (format == AUDIO_FORMAT_PCM_8_BIT) { mCblk->frameSize = mChannelCount * sizeof(int8_t); } else { mCblk->frameSize = sizeof(int8_t); } #endif } } AudioFlinger::RecordThread::RecordTrack::~RecordTrack() { ALOGV("%s", __func__); } // AudioBufferProvider interface status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) { audio_track_cblk_t* cblk = this->cblk(); uint32_t framesAvail; uint32_t framesReq = buffer->frameCount; // Check if last stepServer failed, try to step now if (mStepServerFailed) { if (!step()) goto getNextBuffer_exit; ALOGV("stepServer recovered"); mStepServerFailed = false; } framesAvail = cblk->framesAvailable_l(); if (CC_LIKELY(framesAvail)) { uint32_t s = cblk->server; uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; if (framesReq > framesAvail) { framesReq = framesAvail; } if (framesReq > bufferEnd - s) { framesReq = bufferEnd - s; } buffer->raw = getBuffer(s, framesReq); buffer->frameCount = framesReq; return NO_ERROR; } getNextBuffer_exit: buffer->raw = NULL; buffer->frameCount = 0; return NOT_ENOUGH_DATA; } status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, int triggerSession) { sp thread = mThread.promote(); if (thread != 0) { RecordThread *recordThread = (RecordThread *)thread.get(); return recordThread->start(this, event, triggerSession); } else { return BAD_VALUE; } } void AudioFlinger::RecordThread::RecordTrack::stop() { sp thread = mThread.promote(); if (thread != 0) { RecordThread *recordThread = (RecordThread *)thread.get(); recordThread->mLock.lock(); bool doStop = recordThread->stop_l(this); if (doStop) { TrackBase::reset(); // Force overrun condition to avoid false overrun callback until first data is // read from buffer android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); } recordThread->mLock.unlock(); if (doStop) { AudioSystem::stopInput(recordThread->id()); } } } /*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) { result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User FrameCount\n"); } void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) { snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x %05d\n", (mClient == 0) ? getpid_cached : mClient->pid(), mFormat, mChannelMask, mSessionId, mFrameCount, mState, mCblk->sampleRate, mCblk->server, mCblk->user, mCblk->frameCount); } // ---------------------------------------------------------------------------- AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( PlaybackThread *playbackThread, DuplicatingThread *sourceThread, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount) : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0, IAudioFlinger::TRACK_DEFAULT), mActive(false), mSourceThread(sourceThread) { if (mCblk != NULL) { mCblk->flags |= CBLK_DIRECTION_OUT; mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); mOutBuffer.frameCount = 0; playbackThread->mTracks.add(this); ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); } else { ALOGW("Error creating output track on thread %p", playbackThread); } } AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() { clearBufferQueue(); } status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, int triggerSession) { status_t status = Track::start(event, triggerSession); if (status != NO_ERROR) { return status; } mActive = true; mRetryCount = 127; return status; } void AudioFlinger::PlaybackThread::OutputTrack::stop() { Track::stop(); clearBufferQueue(); mOutBuffer.frameCount = 0; mActive = false; } bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) { Buffer *pInBuffer; Buffer inBuffer; uint32_t channelCount = mChannelCount; bool outputBufferFull = false; inBuffer.frameCount = frames; inBuffer.i16 = data; uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); if (!mActive && frames != 0) { start(); sp thread = mThread.promote(); if (thread != 0) { MixerThread *mixerThread = (MixerThread *)thread.get(); if (mCblk->frameCount > frames){ if (mBufferQueue.size() < kMaxOverFlowBuffers) { uint32_t startFrames = (mCblk->frameCount - frames); pInBuffer = new Buffer; pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; pInBuffer->frameCount = startFrames; pInBuffer->i16 = pInBuffer->mBuffer; memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); mBufferQueue.add(pInBuffer); } else { ALOGW ("OutputTrack::write() %p no more buffers in queue", this); } } } } while (waitTimeLeftMs) { // First write pending buffers, then new data if (mBufferQueue.size()) { pInBuffer = mBufferQueue.itemAt(0); } else { pInBuffer = &inBuffer; } if (pInBuffer->frameCount == 0) { break; } if (mOutBuffer.frameCount == 0) { mOutBuffer.frameCount = pInBuffer->frameCount; nsecs_t startTime = systemTime(); if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); outputBufferFull = true; break; } uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); if (waitTimeLeftMs >= waitTimeMs) { waitTimeLeftMs -= waitTimeMs; } else { waitTimeLeftMs = 0; } } uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); mCblk->stepUser(outFrames); pInBuffer->frameCount -= outFrames; pInBuffer->i16 += outFrames * channelCount; mOutBuffer.frameCount -= outFrames; mOutBuffer.i16 += outFrames * channelCount; if (pInBuffer->frameCount == 0) { if (mBufferQueue.size()) { mBufferQueue.removeAt(0); delete [] pInBuffer->mBuffer; delete pInBuffer; ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); } else { break; } } } // If we could not write all frames, allocate a buffer and queue it for next time. if (inBuffer.frameCount) { sp thread = mThread.promote(); if (thread != 0 && !thread->standby()) { if (mBufferQueue.size() < kMaxOverFlowBuffers) { pInBuffer = new Buffer; pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; pInBuffer->frameCount = inBuffer.frameCount; pInBuffer->i16 = pInBuffer->mBuffer; memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); mBufferQueue.add(pInBuffer); ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); } else { ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); } } } // Calling write() with a 0 length buffer, means that no more data will be written: // If no more buffers are pending, fill output track buffer to make sure it is started // by output mixer. if (frames == 0 && mBufferQueue.size() == 0) { if (mCblk->user < mCblk->frameCount) { frames = mCblk->frameCount - mCblk->user; pInBuffer = new Buffer; pInBuffer->mBuffer = new int16_t[frames * channelCount]; pInBuffer->frameCount = frames; pInBuffer->i16 = pInBuffer->mBuffer; memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); mBufferQueue.add(pInBuffer); } else if (mActive) { stop(); } } return outputBufferFull; } status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) { int active; status_t result; audio_track_cblk_t* cblk = mCblk; uint32_t framesReq = buffer->frameCount; // ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); buffer->frameCount = 0; uint32_t framesAvail = cblk->framesAvailable(); if (framesAvail == 0) { Mutex::Autolock _l(cblk->lock); goto start_loop_here; while (framesAvail == 0) { active = mActive; if (CC_UNLIKELY(!active)) { ALOGV("Not active and NO_MORE_BUFFERS"); return NO_MORE_BUFFERS; } result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); if (result != NO_ERROR) { return NO_MORE_BUFFERS; } // read the server count again start_loop_here: framesAvail = cblk->framesAvailable_l(); } } // if (framesAvail < framesReq) { // return NO_MORE_BUFFERS; // } if (framesReq > framesAvail) { framesReq = framesAvail; } uint32_t u = cblk->user; uint32_t bufferEnd = cblk->userBase + cblk->frameCount; if (framesReq > bufferEnd - u) { framesReq = bufferEnd - u; } buffer->frameCount = framesReq; buffer->raw = (void *)cblk->buffer(u); return NO_ERROR; } void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() { size_t size = mBufferQueue.size(); for (size_t i = 0; i < size; i++) { Buffer *pBuffer = mBufferQueue.itemAt(i); delete [] pBuffer->mBuffer; delete pBuffer; } mBufferQueue.clear(); } // ---------------------------------------------------------------------------- AudioFlinger::Client::Client(const sp& audioFlinger, pid_t pid) : RefBase(), mAudioFlinger(audioFlinger), // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), mPid(pid), mTimedTrackCount(0) { // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer } // Client destructor must be called with AudioFlinger::mLock held AudioFlinger::Client::~Client() { mAudioFlinger->removeClient_l(mPid); } sp AudioFlinger::Client::heap() const { return mMemoryDealer; } // Reserve one of the limited slots for a timed audio track associated // with this client bool AudioFlinger::Client::reserveTimedTrack() { const int kMaxTimedTracksPerClient = 4; Mutex::Autolock _l(mTimedTrackLock); if (mTimedTrackCount >= kMaxTimedTracksPerClient) { ALOGW("can not create timed track - pid %d has exceeded the limit", mPid); return false; } mTimedTrackCount++; return true; } // Release a slot for a timed audio track void AudioFlinger::Client::releaseTimedTrack() { Mutex::Autolock _l(mTimedTrackLock); mTimedTrackCount--; } // ---------------------------------------------------------------------------- AudioFlinger::NotificationClient::NotificationClient(const sp& audioFlinger, const sp& client, sp binder) : mAudioFlinger(audioFlinger), mBinder(binder), mAudioFlingerClient(client) { } AudioFlinger::NotificationClient::~NotificationClient() { } void AudioFlinger::NotificationClient::binderDied(const wp& who) { sp keep(this); mAudioFlinger->removeNotificationClient(mBinder); } // ---------------------------------------------------------------------------- #ifdef QCOM_HARDWARE AudioFlinger::DirectAudioTrack::DirectAudioTrack(const sp& 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) { #ifdef SRS_PROCESSING ALOGD("SRS_Processing - DirectAudioTrack - OutNotify_Init: %p TID %d\n", this, gettid()); POSTPRO_PATCH_ICS_OUTPROC_DIRECT_INIT(this, gettid()); SRS_Processing::ProcessOutRoute(SRS_Processing::AUTO, this, outputDesc->device); #endif if (mFlag & AUDIO_OUTPUT_FLAG_LPA) { createEffectThread(); mAudioFlingerClient = new AudioFlingerDirectTrackClient(this); mAudioFlinger->registerClient(mAudioFlingerClient); allocateBufPool(); #ifdef SRS_PROCESSING } else if (mFlag & AUDIO_OUTPUT_FLAG_TUNNEL) { ALOGV("create effects thread for TUNNEL"); createEffectThread(); mAudioFlingerClient = new AudioFlingerDirectTrackClient(this); mAudioFlinger->registerClient(mAudioFlingerClient); #endif } outputDesc->mVolumeScale = 1.0; mDeathRecipient = new PMDeathRecipient(this); acquireWakeLock(); } AudioFlinger::DirectAudioTrack::~DirectAudioTrack() { #ifdef SRS_PROCESSING ALOGD("SRS_Processing - DirectAudioTrack - OutNotify_Init: %p TID %d\n", this, gettid()); POSTPRO_PATCH_ICS_OUTPROC_DIRECT_EXIT(this, gettid()); #endif if (mFlag & AUDIO_OUTPUT_FLAG_LPA) { requestAndWaitForEffectsThreadExit(); mAudioFlinger->deregisterClient(mAudioFlingerClient); mAudioFlinger->deleteEffectSession(); deallocateBufPool(); #ifdef SRS_PROCESSING } else if (mFlag & AUDIO_OUTPUT_FLAG_TUNNEL) { requestAndWaitForEffectsThreadExit(); mAudioFlinger->deregisterClient(mAudioFlingerClient); mAudioFlinger->deleteEffectSession(); #endif } AudioSystem::releaseOutput(mOutput); releaseWakeLock(); { Mutex::Autolock _l(pmLock); if (mPowerManager != 0) { sp binder = mPowerManager->asBinder(); binder->unlinkToDeath(mDeathRecipient); } } } status_t AudioFlinger::DirectAudioTrack::start() { AudioSystem::startOutput(mOutput, (audio_stream_type_t)mOutputDesc->mStreamType); if(mIsPaused) { mIsPaused = false; mOutputDesc->stream->start(mOutputDesc->stream); } mOutputDesc->mActive = true; 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); } 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); } } 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::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(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) { mEffectsPool.clear(); mEffectsPool = mBufPool; } 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); mOutputDesc->mVolumeLeft = left; mOutputDesc->mVolumeRight = right; mOutputDesc->stream->set_volume(mOutputDesc->stream, left * mOutputDesc->mVolumeScale, right* mOutputDesc->mVolumeScale); } 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 while (!mBufPool.empty()) { List::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); } 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); } void *AudioFlinger::DirectAudioTrack::EffectsThreadWrapper(void *me) { static_cast(me)->EffectsThreadEntry(); return NULL; } void AudioFlinger::DirectAudioTrack::EffectsThreadEntry() { while(1) { mEffectLock.lock(); if (!mEffectConfigChanged && !mKillEffectsThread) { mEffectCv.wait(mEffectLock); } if(mKillEffectsThread) { mEffectLock.unlock(); break; } if (mEffectConfigChanged) { mEffectConfigChanged = false; if (mFlag & AUDIO_OUTPUT_FLAG_LPA) { for ( List::iterator it = mEffectsPool.begin(); it != mEffectsPool.end(); it++) { ALOGV("ete: calling applyEffectsOn buff %x",it->localBuf); bool isEffectsApplied = mAudioFlinger->applyEffectsOn( static_cast(this), (int16_t *)it->localBuf, (int16_t *)mEffectsThreadScratchBuffer, it->bytesToWrite, false); if (isEffectsApplied == true){ ALOGV("ete:dsp updated for local buf %x",it->localBuf); memcpy(it->dspBuf, mEffectsThreadScratchBuffer, it->bytesToWrite); } else ALOGV("ete:dsp updated for local buf %x SKIPPED",it->localBuf); if (mEffectConfigChanged) { ALOGE("ete:effects changed, abort effects application"); break; } } #ifdef SRS_PROCESSING } else if (mFlag & AUDIO_OUTPUT_FLAG_TUNNEL) { ALOGV("applying effects for TUNNEL"); char buffer[2]; //dummy buffer to ensure the SRS processing takes place // The API mandates Sample rate and channel mode. Hence // defaulted the sample rate channel mode to 48000 and 2 respectively POSTPRO_PATCH_ICS_OUTPROC_DIRECT_SAMPLES(static_cast(this), AUDIO_FORMAT_PCM_16_BIT, (int16_t*)buffer, 2, 48000, 2); #endif } } mEffectLock.unlock(); } ALOGV("Effects thread is dead"); mEffectsThreadAlive = false; } void AudioFlinger::DirectAudioTrack::requestAndWaitForEffectsThreadExit() { if (!mEffectsThreadAlive) return; mKillEffectsThread = true; mEffectCv.signal(); pthread_join(mEffectsThread,NULL); ALOGV("effects thread killed"); } void AudioFlinger::DirectAudioTrack::createEffectThread() { //Create the effects thread pthread_attr_t attr; pthread_attr_init(&attr); pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_JOINABLE); mEffectsThreadAlive = true; ALOGV("Creating Effects Thread"); pthread_create(&mEffectsThread, &attr, EffectsThreadWrapper, this); } AudioFlinger::DirectAudioTrack::AudioFlingerDirectTrackClient::AudioFlingerDirectTrackClient(void *obj) { ALOGV("AudioFlinger::DirectAudioTrack::AudioFlingerDirectTrackClient"); pBaseClass = (DirectAudioTrack*)obj; } void AudioFlinger::DirectAudioTrack::AudioFlingerDirectTrackClient::binderDied(const wp& 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 binder = defaultServiceManager()->checkService(String16("power")); if (binder == 0) { ALOGW("Thread %s cannot connect to the power manager service", lockName); } else { mPowerManager = interface_cast(binder); binder->linkToDeath(mDeathRecipient); } } if (mPowerManager != 0 && mWakeLockToken == 0) { sp binder = new BBinder(); status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, binder, String16(lockName)); 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& who) { parentClass->clearPowerManager(); ALOGW("power manager service died !!!"); } #endif // ---------------------------------------------------------------------------- AudioFlinger::TrackHandle::TrackHandle(const sp& track) : BnAudioTrack(), mTrack(track) { } AudioFlinger::TrackHandle::~TrackHandle() { // just stop the track on deletion, associated resources // will be freed from the main thread once all pending buffers have // been played. Unless it's not in the active track list, in which // case we free everything now... mTrack->destroy(); } sp AudioFlinger::TrackHandle::getCblk() const { return mTrack->getCblk(); } status_t AudioFlinger::TrackHandle::start() { return mTrack->start(); } void AudioFlinger::TrackHandle::stop() { mTrack->stop(); } void AudioFlinger::TrackHandle::flush() { mTrack->flush(); } void AudioFlinger::TrackHandle::mute(bool e) { mTrack->mute(e); } void AudioFlinger::TrackHandle::pause() { mTrack->pause(); } status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) { return mTrack->attachAuxEffect(EffectId); } status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, sp* buffer) { if (!mTrack->isTimedTrack()) return INVALID_OPERATION; PlaybackThread::TimedTrack* tt = reinterpret_cast(mTrack.get()); return tt->allocateTimedBuffer(size, buffer); } status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp& buffer, int64_t pts) { if (!mTrack->isTimedTrack()) return INVALID_OPERATION; PlaybackThread::TimedTrack* tt = reinterpret_cast(mTrack.get()); return tt->queueTimedBuffer(buffer, pts); } status_t AudioFlinger::TrackHandle::setMediaTimeTransform( const LinearTransform& xform, int target) { if (!mTrack->isTimedTrack()) return INVALID_OPERATION; PlaybackThread::TimedTrack* tt = reinterpret_cast(mTrack.get()); return tt->setMediaTimeTransform( xform, static_cast(target)); } status_t AudioFlinger::TrackHandle::onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { return BnAudioTrack::onTransact(code, data, reply, flags); } // ---------------------------------------------------------------------------- sp AudioFlinger::openRecord( pid_t pid, audio_io_handle_t input, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, IAudioFlinger::track_flags_t flags, pid_t tid, int *sessionId, status_t *status) { sp recordTrack; sp recordHandle; sp client; #ifdef QCOM_ENHANCED_AUDIO size_t inputBufferSize = 0; uint32_t channelCount = popcount(channelMask); #endif status_t lStatus; RecordThread *thread; size_t inFrameCount; int lSessionId; // check calling permissions if (!recordingAllowed()) { lStatus = PERMISSION_DENIED; goto Exit; } #ifdef QCOM_ENHANCED_AUDIO // Check that audio input stream accepts requested audio parameters inputBufferSize = getInputBufferSize(sampleRate, format, channelCount); if (inputBufferSize == 0) { lStatus = BAD_VALUE; ALOGE("Bad audio input parameters: sampling rate %u, format %d, channels %d", sampleRate, format, channelCount); goto Exit; } #endif // add client to list { // scope for mLock Mutex::Autolock _l(mLock); thread = checkRecordThread_l(input); if (thread == NULL) { lStatus = BAD_VALUE; goto Exit; } client = registerPid_l(pid); // If no audio session id is provided, create one here if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { lSessionId = *sessionId; } else { lSessionId = nextUniqueId(); if (sessionId != NULL) { *sessionId = lSessionId; } } #ifdef QCOM_ENHANCED_AUDIO // frameCount must be a multiple of input buffer size // Change for Codec type uint8_t channelCount = popcount(channelMask); if ((audio_source_t)((int16_t)flags) == AUDIO_SOURCE_VOICE_COMMUNICATION) { inFrameCount = inputBufferSize/channelCount/sizeof(short); } else { if ((format == AUDIO_FORMAT_PCM_16_BIT) || (format == AUDIO_FORMAT_PCM_8_BIT)) { inFrameCount = inputBufferSize/channelCount/sizeof(short); } else if (format == AUDIO_FORMAT_AMR_NB) { inFrameCount = inputBufferSize/channelCount/32; } else if (format == AUDIO_FORMAT_EVRC) { inFrameCount = inputBufferSize/channelCount/23; } else if (format == AUDIO_FORMAT_QCELP) { inFrameCount = inputBufferSize/channelCount/35; } else if (format == AUDIO_FORMAT_AAC) { inFrameCount = inputBufferSize/2048; } else if (format == AUDIO_FORMAT_AMR_WB) { inFrameCount = inputBufferSize/channelCount/61; } } frameCount = ((frameCount - 1)/inFrameCount + 1) * inFrameCount; #endif // create new record track. The record track uses one track in mHardwareMixerThread by convention. recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, frameCount, lSessionId, flags, tid, &lStatus); } if (lStatus != NO_ERROR) { // remove local strong reference to Client before deleting the RecordTrack so that the Client // destructor is called by the TrackBase destructor with mLock held client.clear(); recordTrack.clear(); goto Exit; } // return to handle to client recordHandle = new RecordHandle(recordTrack); lStatus = NO_ERROR; Exit: if (status) { *status = lStatus; } return recordHandle; } // ---------------------------------------------------------------------------- AudioFlinger::RecordHandle::RecordHandle(const sp& recordTrack) : BnAudioRecord(), mRecordTrack(recordTrack) { } AudioFlinger::RecordHandle::~RecordHandle() { stop_nonvirtual(); mRecordTrack->destroy(); } sp AudioFlinger::RecordHandle::getCblk() const { return mRecordTrack->getCblk(); } status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, int triggerSession) { ALOGV("RecordHandle::start()"); return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); } void AudioFlinger::RecordHandle::stop() { stop_nonvirtual(); } void AudioFlinger::RecordHandle::stop_nonvirtual() { ALOGV("RecordHandle::stop()"); mRecordTrack->stop(); } status_t AudioFlinger::RecordHandle::onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { return BnAudioRecord::onTransact(code, data, reply, flags); } // ---------------------------------------------------------------------------- AudioFlinger::RecordThread::RecordThread(const sp& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_io_handle_t id, audio_devices_t device) : ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), // mRsmpInIndex and mInputBytes set by readInputParameters() mReqChannelCount(getInputChannelCount(channelMask)), mReqSampleRate(sampleRate) // mBytesRead is only meaningful while active, and so is cleared in start() // (but might be better to also clear here for dump?) { snprintf(mName, kNameLength, "AudioIn_%X", id); readInputParameters(); } AudioFlinger::RecordThread::~RecordThread() { delete[] mRsmpInBuffer; delete mResampler; delete[] mRsmpOutBuffer; } void AudioFlinger::RecordThread::onFirstRef() { run(mName, PRIORITY_URGENT_AUDIO); } status_t AudioFlinger::RecordThread::readyToRun() { status_t status = initCheck(); ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); return status; } bool AudioFlinger::RecordThread::threadLoop() { AudioBufferProvider::Buffer buffer; sp activeTrack; Vector< sp > effectChains; nsecs_t lastWarning = 0; inputStandBy(); acquireWakeLock(); // used to verify we've read at least once before evaluating how many bytes were read bool readOnce = false; // start recording while (!exitPending()) { processConfigEvents(); { // scope for mLock Mutex::Autolock _l(mLock); checkForNewParameters_l(); if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { standby(); if (exitPending()) break; releaseWakeLock_l(); ALOGV("RecordThread: loop stopping"); // go to sleep mWaitWorkCV.wait(mLock); ALOGV("RecordThread: loop starting"); acquireWakeLock_l(); continue; } if (mActiveTrack != 0) { if (mActiveTrack->mState == TrackBase::PAUSING) { standby(); mActiveTrack.clear(); mStartStopCond.broadcast(); } else if (mActiveTrack->mState == TrackBase::RESUMING) { if (mReqChannelCount != mActiveTrack->channelCount()) { mActiveTrack.clear(); mStartStopCond.broadcast(); } else if (readOnce) { // record start succeeds only if first read from audio input // succeeds if (mBytesRead >= 0) { mActiveTrack->mState = TrackBase::ACTIVE; } else { mActiveTrack.clear(); } mStartStopCond.broadcast(); } mStandby = false; } else if (mActiveTrack->mState == TrackBase::TERMINATED) { removeTrack_l(mActiveTrack); mActiveTrack.clear(); } } lockEffectChains_l(effectChains); } if (mActiveTrack != 0) { if (mActiveTrack->mState != TrackBase::ACTIVE && mActiveTrack->mState != TrackBase::RESUMING) { unlockEffectChains(effectChains); usleep(kRecordThreadSleepUs); continue; } for (size_t i = 0; i < effectChains.size(); i ++) { effectChains[i]->process_l(); } buffer.frameCount = mFrameCount; if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { readOnce = true; size_t framesOut = buffer.frameCount; if (mResampler == NULL) { // no resampling while (framesOut) { size_t framesIn = mFrameCount - mRsmpInIndex; if (framesIn) { int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; if (framesIn > framesOut) framesIn = framesOut; mRsmpInIndex += framesIn; framesOut -= framesIn; if ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT) { memcpy(dst, src, framesIn * mFrameSize); } else { if (mChannelCount == 1) { upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, framesIn); } else { downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, framesIn); } } } if (framesOut && mFrameCount == mRsmpInIndex) { #ifdef QCOM_ENHANCED_AUDIO if (((int) framesOut != mFrameCount) && ((mFormat != AUDIO_FORMAT_PCM_16_BIT)&& ((audio_source_t)mInputSource != AUDIO_SOURCE_VOICE_COMMUNICATION))) { mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, buffer.frameCount * mFrameSize); ALOGV("IR mBytesRead = %d",mBytesRead); if(mBytesRead >= 0 ){ buffer.frameCount = mBytesRead/mFrameSize; } framesOut = 0; } else #endif if (framesOut == mFrameCount && #ifdef QCOM_ENHANCED_AUDIO ((audio_source_t)mInputSource != AUDIO_SOURCE_VOICE_COMMUNICATION) && #endif ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); #ifdef QCOM_ENHANCED_AUDIO if( mBytesRead >= 0 ){ buffer.frameCount = mBytesRead/mFrameSize; } #endif framesOut = 0; } else { mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); mRsmpInIndex = 0; } if (mBytesRead <= 0) { if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { ALOGE("Error reading audio input"); // Force input into standby so that it tries to // recover at next read attempt inputStandBy(); usleep(kRecordThreadSleepUs); } mRsmpInIndex = mFrameCount; framesOut = 0; buffer.frameCount = 0; } } } } else { // resampling memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); // alter output frame count as if we were expecting stereo samples if (mChannelCount == 1 && mReqChannelCount == 1) { framesOut >>= 1; } mResampler->resample(mRsmpOutBuffer, framesOut, this /* AudioBufferProvider* */); // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() // are 32 bit aligned which should be always true. if (mChannelCount == 2 && mReqChannelCount == 1) { ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); // the resampler always outputs stereo samples: do post stereo to mono conversion downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, framesOut); } else { ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); } } if (mFramestoDrop == 0) { mActiveTrack->releaseBuffer(&buffer); } else { if (mFramestoDrop > 0) { mFramestoDrop -= buffer.frameCount; if (mFramestoDrop <= 0) { clearSyncStartEvent(); } } else { mFramestoDrop += buffer.frameCount; if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || mSyncStartEvent->isCancelled()) { ALOGW("Synced record %s, session %d, trigger session %d", (mFramestoDrop >= 0) ? "timed out" : "cancelled", mActiveTrack->sessionId(), (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); clearSyncStartEvent(); } } } mActiveTrack->clearOverflow(); } // client isn't retrieving buffers fast enough else { if (!mActiveTrack->setOverflow()) { nsecs_t now = systemTime(); if ((now - lastWarning) > kWarningThrottleNs) { ALOGW("RecordThread: buffer overflow"); lastWarning = now; } } // Release the processor for a while before asking for a new buffer. // This will give the application more chance to read from the buffer and // clear the overflow. usleep(kRecordThreadSleepUs); } } // enable changes in effect chain unlockEffectChains(effectChains); effectChains.clear(); } standby(); { Mutex::Autolock _l(mLock); mActiveTrack.clear(); mStartStopCond.broadcast(); } releaseWakeLock(); ALOGV("RecordThread %p exiting", this); return false; } void AudioFlinger::RecordThread::standby() { if (!mStandby) { inputStandBy(); mStandby = true; } } void AudioFlinger::RecordThread::inputStandBy() { mInput->stream->common.standby(&mInput->stream->common); } sp AudioFlinger::RecordThread::createRecordTrack_l( const sp& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, int sessionId, IAudioFlinger::track_flags_t flags, pid_t tid, status_t *status) { sp track; status_t lStatus; lStatus = initCheck(); if (lStatus != NO_ERROR) { ALOGE("Audio driver not initialized."); goto Exit; } // FIXME use flags and tid similar to createTrack_l() { // scope for mLock Mutex::Autolock _l(mLock); track = new RecordTrack(this, client, sampleRate, format, channelMask, frameCount, #ifdef QCOM_ENHANCED_AUDIO flags, #endif sessionId); if (track->getCblk() == 0) { lStatus = NO_MEMORY; 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); } lStatus = NO_ERROR; Exit: if (status) { *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 strongMe = this; status_t status = NO_ERROR; if (event == AudioSystem::SYNC_EVENT_NONE) { clearSyncStartEvent(); } else if (event != AudioSystem::SYNC_EVENT_SAME) { mSyncStartEvent = mAudioFlinger->createSyncEvent(event, triggerSession, recordTrack->sessionId(), syncStartEventCallback, this); // 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 (mSyncStartEvent->isCancelled()) { clearSyncStartEvent(); } else { // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); } } { AutoMutex lock(mLock); if (mActiveTrack != 0) { if (recordTrack != mActiveTrack.get()) { status = -EBUSY; } else if (mActiveTrack->mState == TrackBase::PAUSING) { mActiveTrack->mState = TrackBase::ACTIVE; } return status; } recordTrack->mState = TrackBase::IDLE; mActiveTrack = recordTrack; mLock.unlock(); status_t status = AudioSystem::startInput(mId); mLock.lock(); if (status != NO_ERROR) { mActiveTrack.clear(); clearSyncStartEvent(); return status; } mRsmpInIndex = mFrameCount; mBytesRead = 0; if (mResampler != NULL) { mResampler->reset(); } mActiveTrack->mState = TrackBase::RESUMING; // signal thread to start ALOGV("Signal record thread"); mWaitWorkCV.broadcast(); // do not wait for mStartStopCond if exiting if (exitPending()) { mActiveTrack.clear(); status = INVALID_OPERATION; goto startError; } mStartStopCond.wait(mLock); if (mActiveTrack == 0) { ALOGV("Record failed to start"); status = BAD_VALUE; goto startError; } ALOGV("Record started OK"); return status; } startError: AudioSystem::stopInput(mId); clearSyncStartEvent(); return status; } void AudioFlinger::RecordThread::clearSyncStartEvent() { if (mSyncStartEvent != 0) { mSyncStartEvent->cancel(); } mSyncStartEvent.clear(); mFramestoDrop = 0; } void AudioFlinger::RecordThread::syncStartEventCallback(const wp& event) { sp strongEvent = event.promote(); if (strongEvent != 0) { RecordThread *me = (RecordThread *)strongEvent->cookie(); me->handleSyncStartEvent(strongEvent); } } void AudioFlinger::RecordThread::handleSyncStartEvent(const sp& event) { if (event == mSyncStartEvent) { // TODO: use actual buffer filling status instead of 2 buffers when info is available // from audio HAL mFramestoDrop = mFrameCount * 2; } } bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { ALOGV("RecordThread::stop"); if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { return false; } recordTrack->mState = TrackBase::PAUSING; // do not wait for mStartStopCond if exiting if (exitPending()) { return true; } mStartStopCond.wait(mLock); // if we have been restarted, recordTrack == mActiveTrack.get() here if (exitPending() || recordTrack != mActiveTrack.get()) { ALOGV("Record stopped OK"); return true; } return false; } bool AudioFlinger::RecordThread::isValidSyncEvent(const sp& event) const { return false; } status_t AudioFlinger::RecordThread::setSyncEvent(const sp& event) { #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 track = mTracks[i]; if (eventSession == track->sessionId()) { (void) track->setSyncEvent(event); ret = NO_ERROR; } } return ret; #else return BAD_VALUE; #endif } void AudioFlinger::RecordThread::RecordTrack::destroy() { // see comments at AudioFlinger::PlaybackThread::Track::destroy() sp keep(this); { sp thread = mThread.promote(); if (thread != 0) { if (mState == ACTIVE || mState == RESUMING) { AudioSystem::stopInput(thread->id()); } AudioSystem::releaseInput(thread->id()); Mutex::Autolock _l(thread->mLock); RecordThread *recordThread = (RecordThread *) thread.get(); recordThread->destroyTrack_l(this); } } } // destroyTrack_l() must be called with ThreadBase::mLock held void AudioFlinger::RecordThread::destroyTrack_l(const sp& track) { track->mState = TrackBase::TERMINATED; // active tracks are removed by threadLoop() if (mActiveTrack != track) { removeTrack_l(track); } } void AudioFlinger::RecordThread::removeTrack_l(const sp& track) { mTracks.remove(track); // need anything related to effects here? } void AudioFlinger::RecordThread::dump(int fd, const Vector& args) { dumpInternals(fd, args); dumpTracks(fd, args); dumpEffectChains(fd, args); } void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); result.append(buffer); if (mActiveTrack != 0) { snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); result.append(buffer); snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); result.append(buffer); snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); result.append(buffer); snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); result.append(buffer); snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); result.append(buffer); } else { result.append("No active record client\n"); } write(fd, result.string(), result.size()); dumpBase(fd, args); } void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "Input thread %p tracks\n", this); result.append(buffer); RecordTrack::appendDumpHeader(result); for (size_t i = 0; i < mTracks.size(); ++i) { sp track = mTracks[i]; if (track != 0) { track->dump(buffer, SIZE); result.append(buffer); } } if (mActiveTrack != 0) { snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); result.append(buffer); RecordTrack::appendDumpHeader(result); mActiveTrack->dump(buffer, SIZE); result.append(buffer); } write(fd, result.string(), result.size()); } // AudioBufferProvider interface status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) { size_t framesReq = buffer->frameCount; size_t framesReady = mFrameCount - mRsmpInIndex; int channelCount; if (framesReady == 0) { mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); if (mBytesRead <= 0) { if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { ALOGE("RecordThread::getNextBuffer() Error reading audio input"); // Force input into standby so that it tries to // recover at next read attempt inputStandBy(); usleep(kRecordThreadSleepUs); } buffer->raw = NULL; buffer->frameCount = 0; return NOT_ENOUGH_DATA; } mRsmpInIndex = 0; framesReady = mFrameCount; } if (framesReq > framesReady) { framesReq = framesReady; } if (mChannelCount == 1 && mReqChannelCount == 2) { channelCount = 1; } else { channelCount = 2; } buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; buffer->frameCount = framesReq; return NO_ERROR; } // AudioBufferProvider interface void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) { mRsmpInIndex += buffer->frameCount; buffer->frameCount = 0; } bool AudioFlinger::RecordThread::checkForNewParameters_l() { bool reconfig = false; while (!mNewParameters.isEmpty()) { status_t status = NO_ERROR; String8 keyValuePair = mNewParameters[0]; AudioParameter param = AudioParameter(keyValuePair); int value; audio_format_t reqFormat = mFormat; int reqSamplingRate = mReqSampleRate; int reqChannelCount = mReqChannelCount; if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { reqSamplingRate = value; reconfig = true; } if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { reqFormat = (audio_format_t) value; reconfig = true; } if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { reqChannelCount = getInputChannelCount(value); 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 (mActiveTrack != 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 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 && ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && getInputChannelCount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && (reqChannelCount <= FCC_2)) { status = NO_ERROR; } if (status == NO_ERROR) { readInputParameters(); sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); } } } mNewParameters.removeAt(0); mParamStatus = status; mParamCond.signal(); // wait for condition with time out in case the thread calling ThreadBase::setParameters() // already timed out waiting for the status and will never signal the condition. mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); } return reconfig; } String8 AudioFlinger::RecordThread::getParameters(const String8& keys) { char *s; String8 out_s8 = String8(); Mutex::Autolock _l(mLock); if (initCheck() != NO_ERROR) { return out_s8; } s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); out_s8 = String8(s); free(s); return out_s8; } void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { AudioSystem::OutputDescriptor desc; void *param2 = NULL; switch (event) { case AudioSystem::INPUT_OPENED: case AudioSystem::INPUT_CONFIG_CHANGED: desc.channels = mChannelMask; desc.samplingRate = mSampleRate; desc.format = mFormat; desc.frameCount = mFrameCount; desc.latency = 0; param2 = &desc; break; case AudioSystem::INPUT_CLOSED: default: break; } mAudioFlinger->audioConfigChanged_l(event, mId, param2); } void AudioFlinger::RecordThread::readInputParameters() { delete mRsmpInBuffer; // mRsmpInBuffer is always assigned a new[] below delete mRsmpOutBuffer; mRsmpOutBuffer = NULL; delete mResampler; mResampler = NULL; mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); mChannelCount = (uint16_t)getInputChannelCount(mChannelMask); mFormat = mInput->stream->common.get_format(&mInput->stream->common); mFrameSize = audio_stream_frame_size(&mInput->stream->common); mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); mFrameCount = mInputBytes / mFrameSize; mNormalFrameCount = mFrameCount; // not used by record, but used by input effects mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { int channelCount; // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid // stereo to mono post process as the resampler always outputs stereo. if (mChannelCount == 1 && mReqChannelCount == 2) { channelCount = 1; } else { channelCount = 2; } mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); mResampler->setSampleRate(mSampleRate); mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); mRsmpOutBuffer = new int32_t[mFrameCount * 2]; // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples if (mChannelCount == 1 && mReqChannelCount == 1) { mFrameCount >>= 1; } } mRsmpInIndex = mFrameCount; } unsigned int 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 AudioFlinger::RecordThread::sessionIds() const { KeyedVector ids; Mutex::Autolock _l(mLock); for (size_t j = 0; j < mTracks.size(); ++j) { sp 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; } // ---------------------------------------------------------------------------- audio_module_handle_t AudioFlinger::loadHwModule(const char *name) { if (!settingsAllowed()) { return 0; } Mutex::Autolock _l(mLock); return loadHwModule_l(name); } // loadHwModule_l() must be called with AudioFlinger::mLock held audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) { for (size_t i = 0; i < mAudioHwDevs.size(); i++) { if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { ALOGW("loadHwModule() module %s already loaded", name); return mAudioHwDevs.keyAt(i); } } audio_hw_device_t *dev; int rc = load_audio_interface(name, &dev); if (rc) { ALOGI("loadHwModule() error %d loading module %s ", rc, name); return 0; } mHardwareStatus = AUDIO_HW_INIT; rc = dev->init_check(dev); mHardwareStatus = AUDIO_HW_IDLE; if (rc) { ALOGI("loadHwModule() init check error %d for module %s ", rc, name); return 0; } // Check and cache this HAL's level of support for master mute and master // volume. If this is the first HAL opened, and it supports the get // methods, use the initial values provided by the HAL as the current // master mute and volume settings. AudioHwDevice::Flags flags = static_cast(0); { // scope for auto-lock pattern AutoMutex lock(mHardwareLock); #if !defined(ICS_AUDIO_BLOB) && !defined(MR0_AUDIO_BLOB) if (0 == mAudioHwDevs.size()) { mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; if (NULL != dev->get_master_volume) { float mv; if (OK == dev->get_master_volume(dev, &mv)) { mMasterVolume = mv; } } mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; if (NULL != dev->get_master_mute) { bool mm; if (OK == dev->get_master_mute(dev, &mm)) { mMasterMute = mm; } } } #endif mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; if ((NULL != dev->set_master_volume) && (OK == dev->set_master_volume(dev, mMasterVolume))) { flags = static_cast(flags | AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); } #if !defined(ICS_AUDIO_BLOB) && !defined(MR0_AUDIO_BLOB) mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; if ((NULL != dev->set_master_mute) && (OK == dev->set_master_mute(dev, mMasterMute))) { flags = static_cast(flags | AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); } #endif mHardwareStatus = AUDIO_HW_IDLE; } audio_module_handle_t handle = nextUniqueId(); mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", name, dev->common.module->name, dev->common.module->id, handle); return handle; } // ---------------------------------------------------------------------------- int32_t AudioFlinger::getPrimaryOutputSamplingRate() { Mutex::Autolock _l(mLock); PlaybackThread *thread = primaryPlaybackThread_l(); return thread != NULL ? thread->sampleRate() : 0; } int32_t AudioFlinger::getPrimaryOutputFrameCount() { Mutex::Autolock _l(mLock); PlaybackThread *thread = primaryPlaybackThread_l(); return thread != NULL ? thread->frameCountHAL() : 0; } // ---------------------------------------------------------------------------- audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, audio_devices_t *pDevices, uint32_t *pSamplingRate, audio_format_t *pFormat, audio_channel_mask_t *pChannelMask, uint32_t *pLatencyMs, audio_output_flags_t flags) { status_t status; PlaybackThread *thread = NULL; struct audio_config config = { sample_rate: pSamplingRate ? *pSamplingRate : 0, channel_mask: pChannelMask ? *pChannelMask : 0, format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, }; audio_stream_out_t *outStream = NULL; AudioHwDevice *outHwDev; ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", module, (pDevices != NULL) ? *pDevices : 0, config.sample_rate, config.format, config.channel_mask, flags); if (pDevices == NULL || *pDevices == 0) { return 0; } Mutex::Autolock _l(mLock); outHwDev = findSuitableHwDev_l(module, *pDevices); if (outHwDev == NULL) return 0; audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); audio_io_handle_t id = nextUniqueId(); mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; #ifndef ICS_AUDIO_BLOB status = hwDevHal->open_output_stream(hwDevHal, id, *pDevices, (audio_output_flags_t)flags, &config, &outStream); #else status = hwDevHal->open_output_stream(hwDevHal, *pDevices, (int *)&config.format, &config.channel_mask, &config.sample_rate, &outStream); uint32_t newflags = flags | AUDIO_OUTPUT_FLAG_PRIMARY; flags = (audio_output_flags_t)newflags; #endif mHardwareStatus = AUDIO_HW_IDLE; ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", outStream, config.sample_rate, config.format, config.channel_mask, status); if (status == NO_ERROR && outStream != NULL) { AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); #ifdef QCOM_HARDWARE if (flags & AUDIO_OUTPUT_FLAG_LPA || flags & AUDIO_OUTPUT_FLAG_TUNNEL ) { AudioSessionDescriptor *desc = new AudioSessionDescriptor(hwDevHal, outStream, flags); desc->mActive = true; //TODO: no stream type //desc->mStreamType = streamType; desc->mVolumeLeft = 1.0; desc->mVolumeRight = 1.0; desc->device = *pDevices; mDirectAudioTracks.add(id, desc); } else #endif if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || (config.format != AUDIO_FORMAT_PCM_16_BIT) || (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { thread = new DirectOutputThread(this, output, id, *pDevices); ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); } else { thread = new MixerThread(this, output, id, *pDevices); ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); } #ifdef QCOM_HARDWARE if (thread != NULL) #endif mPlaybackThreads.add(id, thread); #ifdef QCOM_HARDWARE // if the device is a A2DP, then this is an A2DP Output if ( true == audio_is_a2dp_device((audio_devices_t) *pDevices) ) { mA2DPHandle = id; ALOGV("A2DP device activated. The handle is set to %d", mA2DPHandle); } #endif if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; if (pFormat != NULL) *pFormat = config.format; if (pChannelMask != NULL) *pChannelMask = config.channel_mask; #ifdef QCOM_HARDWARE if (thread != NULL) { #endif if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); // notify client processes of the new output creation thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); #ifdef QCOM_HARDWARE } else { *pLatencyMs = 0; if ((flags & AUDIO_OUTPUT_FLAG_LPA) || (flags & AUDIO_OUTPUT_FLAG_TUNNEL)) { AudioSessionDescriptor *desc = mDirectAudioTracks.valueFor(id); *pLatencyMs = desc->stream->get_latency(desc->stream); } } #endif // the first primary output opened designates the primary hw device if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { ALOGI("Using module %d has the primary audio interface", module); mPrimaryHardwareDev = outHwDev; #ifdef SRS_PROCESSING SRS_Processing::RawDataSet(NULL, "qdsp hook", &mPrimaryHardwareDev, sizeof(&mPrimaryHardwareDev)); #endif AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_SET_MODE; hwDevHal->set_mode(hwDevHal, mMode); mHardwareStatus = AUDIO_HW_IDLE; } return id; } return 0; } audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2) { Mutex::Autolock _l(mLock); MixerThread *thread1 = checkMixerThread_l(output1); MixerThread *thread2 = checkMixerThread_l(output2); if (thread1 == NULL || thread2 == NULL) { ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); return 0; } audio_io_handle_t id = nextUniqueId(); DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); thread->addOutputTrack(thread2); mPlaybackThreads.add(id, thread); // notify client processes of the new output creation thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); return id; } status_t AudioFlinger::closeOutput(audio_io_handle_t output) { return closeOutput_nonvirtual(output); } status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) { // keep strong reference on the playback thread so that // it is not destroyed while exit() is executed #ifdef QCOM_HARDWARE AudioSessionDescriptor *desc = mDirectAudioTracks.valueFor(output); if (desc) { ALOGV("Closing DirectTrack output %d", output); desc->mActive = false; desc->stream->common.standby(&desc->stream->common); desc->hwDev->close_output_stream(desc->hwDev, desc->stream); desc->trackRefPtr = NULL; mDirectAudioTracks.removeItem(output); audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); delete desc; return NO_ERROR; } #endif sp thread; { Mutex::Autolock _l(mLock); thread = checkPlaybackThread_l(output); if (thread == NULL) { return BAD_VALUE; } ALOGV("closeOutput() %d", output); if (thread->type() == ThreadBase::MIXER) { for (size_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->isDuplicating()) { DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); dupThread->removeOutputTrack((MixerThread *)thread.get()); } } } audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); mPlaybackThreads.removeItem(output); #ifdef QCOM_HARDWARE if (mA2DPHandle == output) { mA2DPHandle = -1; ALOGV("A2DP OutputClosed Notifying Client"); audioConfigChanged_l(AudioSystem::A2DP_OUTPUT_STATE, mA2DPHandle, &mA2DPHandle); } #endif } thread->exit(); // The thread entity (active unit of execution) is no longer running here, // but the ThreadBase container still exists. if (!thread->isDuplicating()) { AudioStreamOut *out = thread->clearOutput(); ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); // from now on thread->mOutput is NULL out->hwDev()->close_output_stream(out->hwDev(), out->stream); delete out; } return NO_ERROR; } status_t AudioFlinger::suspendOutput(audio_io_handle_t output) { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { return BAD_VALUE; } ALOGV("suspendOutput() %d", output); thread->suspend(); return NO_ERROR; } status_t AudioFlinger::restoreOutput(audio_io_handle_t output) { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { return BAD_VALUE; } ALOGV("restoreOutput() %d", output); thread->restore(); return NO_ERROR; } audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, audio_devices_t *pDevices, uint32_t *pSamplingRate, audio_format_t *pFormat, audio_channel_mask_t *pChannelMask) { status_t status; RecordThread *thread = NULL; struct audio_config config = { sample_rate: pSamplingRate ? *pSamplingRate : 0, channel_mask: pChannelMask ? *pChannelMask : 0, format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, }; uint32_t reqSamplingRate = config.sample_rate; audio_format_t reqFormat = config.format; audio_channel_mask_t reqChannels = config.channel_mask; audio_stream_in_t *inStream = NULL; AudioHwDevice *inHwDev; if (pDevices == NULL || *pDevices == 0) { return 0; } Mutex::Autolock _l(mLock); inHwDev = findSuitableHwDev_l(module, *pDevices); if (inHwDev == NULL) return 0; audio_hw_device_t *inHwHal = inHwDev->hwDevice(); audio_io_handle_t id = nextUniqueId(); #ifndef ICS_AUDIO_BLOB status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); #else status = inHwHal->open_input_stream(inHwHal, *pDevices, (int *)&config.format, &config.channel_mask, &config.sample_rate, (audio_in_acoustics_t)0, &inStream); #endif ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", inStream, config.sample_rate, config.format, config.channel_mask, status); // If the input could not be opened with the requested parameters and we can handle the conversion internally, // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo // or stereo to mono conversions on 16 bit PCM inputs. if (status == BAD_VALUE && reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && (config.sample_rate <= 2 * reqSamplingRate) && (getInputChannelCount(config.channel_mask) <= FCC_2) && (getInputChannelCount(reqChannels) <= FCC_2)) { ALOGV("openInput() reopening with proposed sampling rate and channel mask"); inStream = NULL; #ifndef ICS_AUDIO_BLOB status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); #else status = inHwHal->open_input_stream(inHwHal, *pDevices, (int *)&config.format, &config.channel_mask, &config.sample_rate, (audio_in_acoustics_t)0, &inStream); #endif } if (status == NO_ERROR && inStream != NULL) { AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); // Start record thread // RecorThread require both input and output device indication to forward to audio // pre processing modules audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id, device); mRecordThreads.add(id, thread); ALOGV("openInput() created record thread: ID %d thread %p", id, thread); if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; if (pFormat != NULL) *pFormat = config.format; if (pChannelMask != NULL) *pChannelMask = reqChannels; // notify client processes of the new input creation thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); return id; } return 0; } status_t AudioFlinger::closeInput(audio_io_handle_t input) { return closeInput_nonvirtual(input); } status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) { // keep strong reference on the record thread so that // it is not destroyed while exit() is executed sp thread; { Mutex::Autolock _l(mLock); thread = checkRecordThread_l(input); if (thread == 0) { return BAD_VALUE; } ALOGV("closeInput() %d", input); audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); mRecordThreads.removeItem(input); } thread->exit(); // The thread entity (active unit of execution) is no longer running here, // but the ThreadBase container still exists. AudioStreamIn *in = thread->clearInput(); ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); // from now on thread->mInput is NULL in->hwDev()->close_input_stream(in->hwDev(), in->stream); delete in; return NO_ERROR; } status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) { Mutex::Autolock _l(mLock); ALOGV("setStreamOutput() stream %d to output %d", stream, output); for (size_t i = 0; i < mPlaybackThreads.size(); i++) { PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); thread->invalidateTracks(stream); } #ifdef QCOM_HARDWARE if ( mA2DPHandle == output ) { ALOGV("A2DP Activated and hence notifying the client"); audioConfigChanged_l(AudioSystem::A2DP_OUTPUT_STATE, mA2DPHandle, &output); } #endif return NO_ERROR; } int AudioFlinger::newAudioSessionId() { return nextUniqueId(); } void AudioFlinger::acquireAudioSessionId(int audioSession) { Mutex::Autolock _l(mLock); pid_t caller = IPCThreadState::self()->getCallingPid(); ALOGV("acquiring %d from %d", audioSession, caller); size_t num = mAudioSessionRefs.size(); for (size_t i = 0; i< num; i++) { AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); if (ref->mSessionid == audioSession && ref->mPid == caller) { ref->mCnt++; ALOGV(" incremented refcount to %d", ref->mCnt); return; } } mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); ALOGV(" added new entry for %d", audioSession); } void AudioFlinger::releaseAudioSessionId(int audioSession) { Mutex::Autolock _l(mLock); pid_t caller = IPCThreadState::self()->getCallingPid(); ALOGV("releasing %d from %d", audioSession, caller); size_t num = mAudioSessionRefs.size(); for (size_t i = 0; i< num; i++) { AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); if (ref->mSessionid == audioSession && ref->mPid == caller) { ref->mCnt--; ALOGV(" decremented refcount to %d", ref->mCnt); if (ref->mCnt == 0) { mAudioSessionRefs.removeAt(i); delete ref; purgeStaleEffects_l(); } return; } } ALOGW("session id %d not found for pid %d", audioSession, caller); } void AudioFlinger::purgeStaleEffects_l() { ALOGV("purging stale effects"); Vector< sp > chains; for (size_t i = 0; i < mPlaybackThreads.size(); i++) { sp t = mPlaybackThreads.valueAt(i); for (size_t j = 0; j < t->mEffectChains.size(); j++) { sp ec = t->mEffectChains[j]; if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { chains.push(ec); } } } for (size_t i = 0; i < mRecordThreads.size(); i++) { sp t = mRecordThreads.valueAt(i); for (size_t j = 0; j < t->mEffectChains.size(); j++) { sp ec = t->mEffectChains[j]; chains.push(ec); } } for (size_t i = 0; i < chains.size(); i++) { sp ec = chains[i]; int sessionid = ec->sessionId(); sp t = ec->mThread.promote(); if (t == 0) { continue; } size_t numsessionrefs = mAudioSessionRefs.size(); bool found = false; for (size_t k = 0; k < numsessionrefs; k++) { AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); if (ref->mSessionid == sessionid) { ALOGV(" session %d still exists for %d with %d refs", sessionid, ref->mPid, ref->mCnt); found = true; break; } } if (!found) { Mutex::Autolock _l (t->mLock); // remove all effects from the chain while (ec->mEffects.size()) { sp effect = ec->mEffects[0]; effect->unPin(); t->removeEffect_l(effect); if (effect->purgeHandles()) { t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); } AudioSystem::unregisterEffect(effect->id()); } } } return; } // checkPlaybackThread_l() must be called with AudioFlinger::mLock held AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const { return mPlaybackThreads.valueFor(output).get(); } // checkMixerThread_l() must be called with AudioFlinger::mLock held AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const { PlaybackThread *thread = checkPlaybackThread_l(output); return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; } // checkRecordThread_l() must be called with AudioFlinger::mLock held AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const { return mRecordThreads.valueFor(input).get(); } uint32_t AudioFlinger::nextUniqueId() { return android_atomic_inc(&mNextUniqueId); } AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const { for (size_t i = 0; i < mPlaybackThreads.size(); i++) { PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); if(thread->isDuplicating()) { continue; } AudioStreamOut *output = thread->getOutput(); if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { return thread; } } return NULL; } audio_devices_t AudioFlinger::primaryOutputDevice_l() const { PlaybackThread *thread = primaryPlaybackThread_l(); if (thread == NULL) { return 0; } return thread->outDevice(); } sp AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, int triggerSession, int listenerSession, sync_event_callback_t callBack, void *cookie) { Mutex::Autolock _l(mLock); sp event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); status_t playStatus = NAME_NOT_FOUND; status_t recStatus = NAME_NOT_FOUND; for (size_t i = 0; i < mPlaybackThreads.size(); i++) { playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); if (playStatus == NO_ERROR) { return event; } } for (size_t i = 0; i < mRecordThreads.size(); i++) { recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); if (recStatus == NO_ERROR) { return event; } } if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { mPendingSyncEvents.add(event); } else { ALOGV("createSyncEvent() invalid event %d", event->type()); event.clear(); } return event; } // ---------------------------------------------------------------------------- // Effect management // ---------------------------------------------------------------------------- status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const { Mutex::Autolock _l(mLock); return EffectQueryNumberEffects(numEffects); } status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const { Mutex::Autolock _l(mLock); return EffectQueryEffect(index, descriptor); } status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, effect_descriptor_t *descriptor) const { Mutex::Autolock _l(mLock); return EffectGetDescriptor(pUuid, descriptor); } sp AudioFlinger::createEffect(pid_t pid, effect_descriptor_t *pDesc, const sp& effectClient, int32_t priority, audio_io_handle_t io, int sessionId, status_t *status, int *id, int *enabled) { status_t lStatus = NO_ERROR; sp handle; effect_descriptor_t desc; ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", pid, effectClient.get(), priority, sessionId, io); if (pDesc == NULL) { lStatus = BAD_VALUE; goto Exit; } // check audio settings permission for global effects if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { lStatus = PERMISSION_DENIED; goto Exit; } // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects // that can only be created by audio policy manager (running in same process) if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { lStatus = PERMISSION_DENIED; goto Exit; } if (io == 0) { if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { // output must be specified by AudioPolicyManager when using session // AUDIO_SESSION_OUTPUT_STAGE lStatus = BAD_VALUE; goto Exit; } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { // if the output returned by getOutputForEffect() is removed before we lock the // mutex below, the call to checkPlaybackThread_l(io) below will detect it // and we will exit safely io = AudioSystem::getOutputForEffect(&desc); } } { Mutex::Autolock _l(mLock); if (!EffectIsNullUuid(&pDesc->uuid)) { // if uuid is specified, request effect descriptor lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); if (lStatus < 0) { ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); goto Exit; } } else { // if uuid is not specified, look for an available implementation // of the required type in effect factory if (EffectIsNullUuid(&pDesc->type)) { ALOGW("createEffect() no effect type"); lStatus = BAD_VALUE; goto Exit; } uint32_t numEffects = 0; effect_descriptor_t d; d.flags = 0; // prevent compiler warning bool found = false; lStatus = EffectQueryNumberEffects(&numEffects); if (lStatus < 0) { ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); goto Exit; } for (uint32_t i = 0; i < numEffects; i++) { lStatus = EffectQueryEffect(i, &desc); if (lStatus < 0) { ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); continue; } if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { // If matching type found save effect descriptor. If the session is // 0 and the effect is not auxiliary, continue enumeration in case // an auxiliary version of this effect type is available found = true; d = desc; if (sessionId != AUDIO_SESSION_OUTPUT_MIX || (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { break; } } } if (!found) { lStatus = BAD_VALUE; ALOGW("createEffect() effect not found"); goto Exit; } // For same effect type, chose auxiliary version over insert version if // connect to output mix (Compliance to OpenSL ES) if (sessionId == AUDIO_SESSION_OUTPUT_MIX && (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { desc = d; } } // Do not allow auxiliary effects on a session different from 0 (output mix) if (sessionId != AUDIO_SESSION_OUTPUT_MIX && (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { lStatus = INVALID_OPERATION; goto Exit; } // check recording permission for visualizer if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && !recordingAllowed()) { lStatus = PERMISSION_DENIED; goto Exit; } // return effect descriptor *pDesc = desc; // If output is not specified try to find a matching audio session ID in one of the // output threads. // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX // because of code checking output when entering the function. // Note: io is never 0 when creating an effect on an input if (io == 0) { // look for the thread where the specified audio session is present for (size_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { io = mPlaybackThreads.keyAt(i); break; } } if (io == 0) { for (size_t i = 0; i < mRecordThreads.size(); i++) { if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { io = mRecordThreads.keyAt(i); break; } } } // If no output thread contains the requested session ID, default to // first output. The effect chain will be moved to the correct output // thread when a track with the same session ID is created if (io == 0 && mPlaybackThreads.size()) { io = mPlaybackThreads.keyAt(0); } ALOGV("createEffect() got io %d for effect %s", io, desc.name); } ThreadBase *thread = checkRecordThread_l(io); if (thread == NULL) { thread = checkPlaybackThread_l(io); if (thread == NULL) { ALOGE("createEffect() unknown output thread"); lStatus = BAD_VALUE; goto Exit; } } sp client = registerPid_l(pid); // create effect on selected output thread handle = thread->createEffect_l(client, effectClient, priority, sessionId, &desc, enabled, &lStatus); if (handle != 0 && id != NULL) { *id = handle->id(); } } Exit: if (status != NULL) { *status = lStatus; } return handle; } status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, audio_io_handle_t dstOutput) { ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", sessionId, srcOutput, dstOutput); Mutex::Autolock _l(mLock); if (srcOutput == dstOutput) { ALOGW("moveEffects() same dst and src outputs %d", dstOutput); return NO_ERROR; } PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); if (srcThread == NULL) { ALOGW("moveEffects() bad srcOutput %d", srcOutput); return BAD_VALUE; } PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); if (dstThread == NULL) { ALOGW("moveEffects() bad dstOutput %d", dstOutput); return BAD_VALUE; } Mutex::Autolock _dl(dstThread->mLock); Mutex::Autolock _sl(srcThread->mLock); moveEffectChain_l(sessionId, srcThread, dstThread, false); return NO_ERROR; } // moveEffectChain_l must be called with both srcThread and dstThread mLocks held status_t AudioFlinger::moveEffectChain_l(int sessionId, AudioFlinger::PlaybackThread *srcThread, AudioFlinger::PlaybackThread *dstThread, bool reRegister) { ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", sessionId, srcThread, dstThread); sp chain = srcThread->getEffectChain_l(sessionId); if (chain == 0) { ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", sessionId, srcThread); return INVALID_OPERATION; } // remove chain first. This is useful only if reconfiguring effect chain on same output thread, // so that a new chain is created with correct parameters when first effect is added. This is // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is // removed. srcThread->removeEffectChain_l(chain); // transfer all effects one by one so that new effect chain is created on new thread with // correct buffer sizes and audio parameters and effect engines reconfigured accordingly audio_io_handle_t dstOutput = dstThread->id(); sp dstChain; uint32_t strategy = 0; // prevent compiler warning sp effect = chain->getEffectFromId_l(0); while (effect != 0) { srcThread->removeEffect_l(effect); dstThread->addEffect_l(effect); // removeEffect_l() has stopped the effect if it was active so it must be restarted if (effect->state() == EffectModule::ACTIVE || effect->state() == EffectModule::STOPPING) { effect->start(); } // if the move request is not received from audio policy manager, the effect must be // re-registered with the new strategy and output if (dstChain == 0) { dstChain = effect->chain().promote(); if (dstChain == 0) { ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); srcThread->addEffect_l(effect); return NO_INIT; } strategy = dstChain->strategy(); } if (reRegister) { AudioSystem::unregisterEffect(effect->id()); AudioSystem::registerEffect(&effect->desc(), dstOutput, strategy, sessionId, effect->id()); } effect = chain->getEffectFromId_l(0); } return NO_ERROR; } // PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held sp AudioFlinger::ThreadBase::createEffect_l( const sp& client, const sp& effectClient, int32_t priority, int sessionId, effect_descriptor_t *desc, int *enabled, status_t *status ) { sp effect; sp handle; status_t lStatus; sp 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; } // Do not allow effects with session ID 0 on direct output or duplicating threads // TODO: add rule for hw accelerated effects on direct outputs with non PCM format if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", desc->name, sessionId); 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_HARDWARE 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; } lStatus = chain->addEffect_l(effect); if (lStatus != NO_ERROR) { goto Exit; } effectCreated = true; effect->setDevice(mOutDevice); effect->setDevice(mInDevice); effect->setMode(mAudioFlinger->getMode()); effect->setAudioSource(mAudioSource); #ifdef QCOM_HARDWARE if(chain == mAudioFlinger->mLPAEffectChain) { effect->setLPAFlag(true); } #endif } // create effect handle and connect it to effect module handle = new EffectHandle(effect, client, effectClient, priority); 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(); } if (status != NULL) { *status = lStatus; } return handle; } sp AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) { Mutex::Autolock _l(mLock); return getEffect_l(sessionId, effectId); } sp AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) { sp 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& effect) { // check for existing effect chain with the requested audio session int sessionId = effect->sessionId(); sp chain = getEffectChain_l(sessionId); bool chainCreated = false; 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; } 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& 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 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 >& effectChains) { effectChains = mEffectChains; #ifdef QCOM_HARDWARE mAudioFlinger->mAllChainsLocked = true; #endif for (size_t i = 0; i < mEffectChains.size(); i++) { #ifdef QCOM_HARDWARE if (mEffectChains[i] != mAudioFlinger->mLPAEffectChain) { #endif mEffectChains[i]->lock(); #ifdef QCOM_HARDWARE } else { mAudioFlinger-> mAllChainsLocked = false; } #endif } } void AudioFlinger::ThreadBase::unlockEffectChains( const Vector< sp >& effectChains) { for (size_t i = 0; i < effectChains.size(); i++) { #ifdef QCOM_HARDWARE if (mAudioFlinger-> mAllChainsLocked || mEffectChains[i] != mAudioFlinger->mLPAEffectChain) #endif effectChains[i]->unlock(); } } sp AudioFlinger::ThreadBase::getEffectChain(int sessionId) { Mutex::Autolock _l(mLock); return getEffectChain_l(sessionId); } sp 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::disconnectEffect(const sp& effect, EffectHandle *handle, bool unpinIfLast) { Mutex::Autolock _l(mLock); ALOGV("disconnectEffect() %p effect %p", this, effect.get()); // delete the effect module if removing last handle on it if (effect->removeHandle(handle) == 0) { if (!effect->isPinned() || unpinIfLast) { removeEffect_l(effect); AudioSystem::unregisterEffect(effect->id()); } } } status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp& chain) { int session = chain->sessionId(); int16_t *buffer = mMixBuffer; 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 mix 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 = 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 = 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->setInBuffer(buffer, ownsBuffer); chain->setOutBuffer(mMixBuffer); // 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& 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 = 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 = mTracks[i]; if (session == track->sessionId()) { track->setMainBuffer(mMixBuffer); chain->decTrackCnt(); } } break; } } return mEffectChains.size(); } status_t AudioFlinger::PlaybackThread::attachAuxEffect( const sp track, int EffectId) { Mutex::Autolock _l(mLock); return attachAuxEffect_l(track, EffectId); } status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( const sp 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 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 = mTracks[i]; if (track->auxEffectId() == effectId) { attachAuxEffect_l(track, 0); } } } status_t AudioFlinger::RecordThread::addEffectChain_l(const sp& chain) { // only one chain per input thread if (mEffectChains.size() != 0) { return INVALID_OPERATION; } ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); chain->setInBuffer(NULL); chain->setOutBuffer(NULL); checkSuspendOnAddEffectChain_l(chain); mEffectChains.add(chain); return NO_ERROR; } size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp& 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; } // ---------------------------------------------------------------------------- // EffectModule implementation // ---------------------------------------------------------------------------- #undef LOG_TAG #define LOG_TAG "AudioFlinger::EffectModule" AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, const wp& chain, effect_descriptor_t *desc, int id, int sessionId) : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mDescriptor(*desc), // mConfig is set by configure() and not used before then mEffectInterface(NULL), mStatus(NO_INIT), mState(IDLE), // mMaxDisableWaitCnt is set by configure() and not used before then // mDisableWaitCnt is set by process() and updateState() and not used before then mSuspended(false) #ifdef QCOM_HARDWARE ,mIsForLPA(false) #endif { ALOGV("Constructor %p", this); int lStatus; // create effect engine from effect factory mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); if (mStatus != NO_ERROR) { return; } lStatus = init(); if (lStatus < 0) { mStatus = lStatus; goto Error; } ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); return; Error: EffectRelease(mEffectInterface); mEffectInterface = NULL; ALOGV("Constructor Error %d", mStatus); } AudioFlinger::EffectModule::~EffectModule() { ALOGV("Destructor %p", this); if (mEffectInterface != NULL) { if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { sp thread = mThread.promote(); if (thread != 0) { audio_stream_t *stream = thread->stream(); if (stream != NULL) { stream->remove_audio_effect(stream, mEffectInterface); } } } // release effect engine EffectRelease(mEffectInterface); } } status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) { status_t status; Mutex::Autolock _l(mLock); int priority = handle->priority(); size_t size = mHandles.size(); EffectHandle *controlHandle = NULL; size_t i; for (i = 0; i < size; i++) { EffectHandle *h = mHandles[i]; if (h == NULL || h->destroyed_l()) continue; // first non destroyed handle is considered in control if (controlHandle == NULL) controlHandle = h; if (h->priority() <= priority) break; } // if inserted in first place, move effect control from previous owner to this handle if (i == 0) { bool enabled = false; if (controlHandle != NULL) { enabled = controlHandle->enabled(); controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); } handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); status = NO_ERROR; } else { status = ALREADY_EXISTS; } ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); mHandles.insertAt(handle, i); return status; } size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) { Mutex::Autolock _l(mLock); size_t size = mHandles.size(); size_t i; for (i = 0; i < size; i++) { if (mHandles[i] == handle) break; } if (i == size) { return size; } ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); mHandles.removeAt(i); // if removed from first place, move effect control from this handle to next in line if (i == 0) { EffectHandle *h = controlHandle_l(); if (h != NULL) { h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); } } // Prevent calls to process() and other functions on effect interface from now on. // The effect engine will be released by the destructor when the last strong reference on // this object is released which can happen after next process is called. if (mHandles.size() == 0 && !mPinned) { mState = DESTROYED; } return mHandles.size(); } // must be called with EffectModule::mLock held AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() { // the first valid handle in the list has control over the module for (size_t i = 0; i < mHandles.size(); i++) { EffectHandle *h = mHandles[i]; if (h != NULL && !h->destroyed_l()) { return h; } } return NULL; } size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) { #ifdef QCOM_HARDWARE setEnabled(false); #endif ALOGV("disconnect() %p handle %p", this, handle); // keep a strong reference on this EffectModule to avoid calling the // destructor before we exit sp keep(this); { sp thread = mThread.promote(); if (thread != 0) { thread->disconnectEffect(keep, handle, unpinIfLast); } } return mHandles.size(); } void AudioFlinger::EffectModule::updateState() { Mutex::Autolock _l(mLock); switch (mState) { case RESTART: reset_l(); // FALL THROUGH case STARTING: // clear auxiliary effect input buffer for next accumulation if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { memset(mConfig.inputCfg.buffer.raw, 0, mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); } start_l(); mState = ACTIVE; break; case STOPPING: stop_l(); mDisableWaitCnt = mMaxDisableWaitCnt; mState = STOPPED; break; case STOPPED: // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the // turn off sequence. if (--mDisableWaitCnt == 0) { reset_l(); mState = IDLE; } break; default: //IDLE , ACTIVE, DESTROYED break; } } void AudioFlinger::EffectModule::process() { Mutex::Autolock _l(mLock); if (mState == DESTROYED || mEffectInterface == NULL || mConfig.inputCfg.buffer.raw == NULL || mConfig.outputCfg.buffer.raw == NULL) { return; } if (isProcessEnabled()) { // do 32 bit to 16 bit conversion for auxiliary effect input buffer if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { ditherAndClamp(mConfig.inputCfg.buffer.s32, mConfig.inputCfg.buffer.s32, mConfig.inputCfg.buffer.frameCount/2); } // do the actual processing in the effect engine int ret = (*mEffectInterface)->process(mEffectInterface, &mConfig.inputCfg.buffer, &mConfig.outputCfg.buffer); // force transition to IDLE state when engine is ready if (mState == STOPPED && ret == -ENODATA) { mDisableWaitCnt = 1; } // clear auxiliary effect input buffer for next accumulation if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { memset(mConfig.inputCfg.buffer.raw, 0, mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); } } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { // If an insert effect is idle and input buffer is different from output buffer, // accumulate input onto output sp chain = mChain.promote(); if (chain != 0 && chain->activeTrackCnt() != 0) { size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here int16_t *in = mConfig.inputCfg.buffer.s16; int16_t *out = mConfig.outputCfg.buffer.s16; for (size_t i = 0; i < frameCnt; i++) { out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); } } } } void AudioFlinger::EffectModule::reset_l() { if (mEffectInterface == NULL) { return; } (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); } #ifndef QCOM_HARDWARE status_t AudioFlinger::EffectModule::configure() { #else status_t AudioFlinger::EffectModule::configure(bool isForLPA, int sampleRate, int channelCount, int frameCount) { uint32_t channels; // Acquire lock here to make sure that any other thread does not delete // the effect handle and release the effect module. Mutex::Autolock _l(mLock); #endif if (mEffectInterface == NULL) { return NO_INIT; } sp thread = mThread.promote(); if (thread == 0) { return DEAD_OBJECT; } // TODO: handle configuration of effects replacing track process audio_channel_mask_t channelMask = thread->channelMask(); #ifdef QCOM_HARDWARE mIsForLPA = isForLPA; if(isForLPA) { if (channelCount == 1) { channels = AUDIO_CHANNEL_OUT_MONO; } else { channels = AUDIO_CHANNEL_OUT_STEREO; } // ALOGV("%s: LPA ON - channels %d", __func__, channels); } else { if (thread->channelCount() == 1) { channels = AUDIO_CHANNEL_OUT_MONO; } else { channels = AUDIO_CHANNEL_OUT_STEREO; } } #endif if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; } else { mConfig.inputCfg.channels = channelMask; } mConfig.outputCfg.channels = channelMask; mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; #ifdef QCOM_HARDWARE if(isForLPA){ mConfig.inputCfg.samplingRate = sampleRate; ALOGV("%s: LPA ON - sampleRate %d", __func__, sampleRate); } else #endif mConfig.inputCfg.samplingRate = thread->sampleRate(); mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; mConfig.inputCfg.bufferProvider.cookie = NULL; mConfig.inputCfg.bufferProvider.getBuffer = NULL; mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; mConfig.outputCfg.bufferProvider.cookie = NULL; mConfig.outputCfg.bufferProvider.getBuffer = NULL; mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; // Insert effect: // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, // always overwrites output buffer: input buffer == output buffer // - in other sessions: // last effect in the chain accumulates in output buffer: input buffer != output buffer // other effect: overwrites output buffer: input buffer == output buffer // Auxiliary effect: // accumulates in output buffer: input buffer != output buffer // Therefore: accumulate <=> input buffer != output buffer if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; } else { mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; } mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; #ifdef QCOM_HARDWARE if(isForLPA) { mConfig.inputCfg.buffer.frameCount = frameCount; ALOGV("%s: LPA ON - frameCount %d", __func__, frameCount); } else #endif mConfig.inputCfg.buffer.frameCount = thread->frameCount(); mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; // ALOGV("configure() %p thread %p buffer %p framecount %d", // this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); status_t cmdStatus; uint32_t size = sizeof(int); status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_CONFIG, sizeof(effect_config_t), &mConfig, &size, &cmdStatus); if (status == 0) { status = cmdStatus; } if (status == 0 && (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; effect_param_t *p = (effect_param_t *)buf32; p->psize = sizeof(uint32_t); p->vsize = sizeof(uint32_t); size = sizeof(int); *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; uint32_t latency = 0; PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); if (pbt != NULL) { latency = pbt->latency_l(); } *((int32_t *)p->data + 1)= latency; (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_PARAM, sizeof(effect_param_t) + 8, &buf32, &size, &cmdStatus); } mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / (1000 * mConfig.outputCfg.buffer.frameCount); return status; } status_t AudioFlinger::EffectModule::init() { Mutex::Autolock _l(mLock); if (mEffectInterface == NULL) { return NO_INIT; } status_t cmdStatus; uint32_t size = sizeof(status_t); status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_INIT, 0, NULL, &size, &cmdStatus); if (status == 0) { status = cmdStatus; } return status; } status_t AudioFlinger::EffectModule::start() { Mutex::Autolock _l(mLock); return start_l(); } status_t AudioFlinger::EffectModule::start_l() { if (mEffectInterface == NULL) { return NO_INIT; } status_t cmdStatus; uint32_t size = sizeof(status_t); status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_ENABLE, 0, NULL, &size, &cmdStatus); if (status == 0) { status = cmdStatus; } if (status == 0 && ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { sp thread = mThread.promote(); if (thread != 0) { audio_stream_t *stream = thread->stream(); if (stream != NULL) { stream->add_audio_effect(stream, mEffectInterface); } } } return status; } status_t AudioFlinger::EffectModule::stop() { Mutex::Autolock _l(mLock); return stop_l(); } status_t AudioFlinger::EffectModule::stop_l() { if (mEffectInterface == NULL) { return NO_INIT; } status_t cmdStatus; uint32_t size = sizeof(status_t); status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_DISABLE, 0, NULL, &size, &cmdStatus); if (status == 0) { status = cmdStatus; } if (status == 0 && ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { sp thread = mThread.promote(); if (thread != 0) { audio_stream_t *stream = thread->stream(); if (stream != NULL) { stream->remove_audio_effect(stream, mEffectInterface); } } } return status; } status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, uint32_t cmdSize, void *pCmdData, uint32_t *replySize, void *pReplyData) { Mutex::Autolock _l(mLock); // ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); if (mState == DESTROYED || mEffectInterface == NULL) { return NO_INIT; } status_t status = (*mEffectInterface)->command(mEffectInterface, cmdCode, cmdSize, pCmdData, replySize, pReplyData); if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { uint32_t size = (replySize == NULL) ? 0 : *replySize; for (size_t i = 1; i < mHandles.size(); i++) { EffectHandle *h = mHandles[i]; if (h != NULL && !h->destroyed_l()) { h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); } } } return status; } status_t AudioFlinger::EffectModule::setEnabled(bool enabled) { Mutex::Autolock _l(mLock); return setEnabled_l(enabled); } // must be called with EffectModule::mLock held status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) { #ifdef QCOM_HARDWARE bool effectStateChanged = false; #endif ALOGV("setEnabled %p enabled %d", this, enabled); if (enabled != isEnabled()) { #ifdef QCOM_HARDWARE effectStateChanged = true; #endif status_t status = AudioSystem::setEffectEnabled(mId, enabled); if (enabled && status != NO_ERROR) { return status; } switch (mState) { // going from disabled to enabled case IDLE: mState = STARTING; break; case STOPPED: mState = RESTART; break; case STOPPING: mState = ACTIVE; break; // going from enabled to disabled case RESTART: mState = STOPPED; break; case STARTING: mState = IDLE; break; case ACTIVE: mState = STOPPING; break; case DESTROYED: return NO_ERROR; // simply ignore as we are being destroyed } for (size_t i = 1; i < mHandles.size(); i++) { EffectHandle *h = mHandles[i]; if (h != NULL && !h->destroyed_l()) { h->setEnabled(enabled); } } } #ifdef QCOM_HARDWARE /* Send notification event to LPA Player when an effect for LPA output is enabled or disabled. */ if (effectStateChanged && mIsForLPA) { sp thread = mThread.promote(); thread->effectConfigChanged(); } #endif return NO_ERROR; } bool AudioFlinger::EffectModule::isEnabled() const { switch (mState) { case RESTART: case STARTING: case ACTIVE: return true; case IDLE: case STOPPING: case STOPPED: case DESTROYED: default: return false; } } bool AudioFlinger::EffectModule::isProcessEnabled() const { switch (mState) { case RESTART: case ACTIVE: case STOPPING: case STOPPED: return true; case IDLE: case STARTING: case DESTROYED: default: return false; } } status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) { Mutex::Autolock _l(mLock); status_t status = NO_ERROR; // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) if (isProcessEnabled() && ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { status_t cmdStatus; uint32_t volume[2]; uint32_t *pVolume = NULL; uint32_t size = sizeof(volume); volume[0] = *left; volume[1] = *right; if (controller) { pVolume = volume; } status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_VOLUME, size, volume, &size, pVolume); if (controller && status == NO_ERROR && size == sizeof(volume)) { *left = volume[0]; *right = volume[1]; } } return status; } status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) { if (device == AUDIO_DEVICE_NONE) { return NO_ERROR; } Mutex::Autolock _l(mLock); status_t status = NO_ERROR; if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { status_t cmdStatus; uint32_t size = sizeof(status_t); uint32_t cmd = audio_is_output_devices(device) ? EFFECT_CMD_SET_DEVICE : EFFECT_CMD_SET_INPUT_DEVICE; status = (*mEffectInterface)->command(mEffectInterface, cmd, sizeof(uint32_t), &device, &size, &cmdStatus); } return status; } status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) { Mutex::Autolock _l(mLock); status_t status = NO_ERROR; if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { status_t cmdStatus; uint32_t size = sizeof(status_t); status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_AUDIO_MODE, sizeof(audio_mode_t), &mode, &size, &cmdStatus); if (status == NO_ERROR) { status = cmdStatus; } } return status; } status_t AudioFlinger::EffectModule::setAudioSource(audio_source_t source) { Mutex::Autolock _l(mLock); status_t status = NO_ERROR; if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_SOURCE_MASK) == EFFECT_FLAG_AUDIO_SOURCE_IND) { uint32_t size = 0; status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_AUDIO_SOURCE, sizeof(audio_source_t), &source, &size, NULL); } return status; } void AudioFlinger::EffectModule::setSuspended(bool suspended) { Mutex::Autolock _l(mLock); mSuspended = suspended; } bool AudioFlinger::EffectModule::suspended() const { Mutex::Autolock _l(mLock); return mSuspended; } bool AudioFlinger::EffectModule::purgeHandles() { bool enabled = false; Mutex::Autolock _l(mLock); for (size_t i = 0; i < mHandles.size(); i++) { EffectHandle *handle = mHandles[i]; if (handle != NULL && !handle->destroyed_l()) { handle->effect().clear(); if (handle->hasControl()) { enabled = handle->enabled(); } } } return enabled; } void AudioFlinger::EffectModule::dump(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); result.append(buffer); bool locked = tryLock(mLock); // failed to lock - AudioFlinger is probably deadlocked if (!locked) { result.append("\t\tCould not lock Fx mutex:\n"); } result.append("\t\tSession Status State Engine:\n"); snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", mSessionId, mStatus, mState, (uint32_t)mEffectInterface); result.append(buffer); result.append("\t\tDescriptor:\n"); snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); result.append(buffer); snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); result.append(buffer); snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", mDescriptor.apiVersion, mDescriptor.flags); result.append(buffer); snprintf(buffer, SIZE, "\t\t- name: %s\n", mDescriptor.name); result.append(buffer); snprintf(buffer, SIZE, "\t\t- implementor: %s\n", mDescriptor.implementor); result.append(buffer); result.append("\t\t- Input configuration:\n"); result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", (uint32_t)mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount, mConfig.inputCfg.samplingRate, mConfig.inputCfg.channels, mConfig.inputCfg.format); result.append(buffer); result.append("\t\t- Output configuration:\n"); result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", (uint32_t)mConfig.outputCfg.buffer.raw, mConfig.outputCfg.buffer.frameCount, mConfig.outputCfg.samplingRate, mConfig.outputCfg.channels, mConfig.outputCfg.format); result.append(buffer); snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); result.append(buffer); result.append("\t\t\tPid Priority Ctrl Locked client server\n"); for (size_t i = 0; i < mHandles.size(); ++i) { EffectHandle *handle = mHandles[i]; if (handle != NULL && !handle->destroyed_l()) { handle->dump(buffer, SIZE); result.append(buffer); } } result.append("\n"); write(fd, result.string(), result.length()); if (locked) { mLock.unlock(); } } // ---------------------------------------------------------------------------- // EffectHandle implementation // ---------------------------------------------------------------------------- #undef LOG_TAG #define LOG_TAG "AudioFlinger::EffectHandle" AudioFlinger::EffectHandle::EffectHandle(const sp& effect, const sp& client, const sp& effectClient, int32_t priority) : BnEffect(), mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) { ALOGV("constructor %p", this); if (client == 0) { return; } int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); if (mCblkMemory != 0) { mCblk = static_cast(mCblkMemory->pointer()); if (mCblk != NULL) { new(mCblk) effect_param_cblk_t(); mBuffer = (uint8_t *)mCblk + bufOffset; } } else { ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); return; } } AudioFlinger::EffectHandle::~EffectHandle() { ALOGV("Destructor %p", this); if (mEffect == 0) { mDestroyed = true; return; } mEffect->lock(); mDestroyed = true; mEffect->unlock(); disconnect(false); } status_t AudioFlinger::EffectHandle::enable() { ALOGV("enable %p", this); if (!mHasControl) return INVALID_OPERATION; if (mEffect == 0) return DEAD_OBJECT; if (mEnabled) { return NO_ERROR; } mEnabled = true; sp thread = mEffect->thread().promote(); if (thread != 0) { thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); } // checkSuspendOnEffectEnabled() can suspend this same effect when enabled if (mEffect->suspended()) { return NO_ERROR; } status_t status = mEffect->setEnabled(true); if (status != NO_ERROR) { if (thread != 0) { thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); } mEnabled = false; } return status; } status_t AudioFlinger::EffectHandle::disable() { ALOGV("disable %p", this); if (!mHasControl) return INVALID_OPERATION; if (mEffect == 0) return DEAD_OBJECT; if (!mEnabled) { return NO_ERROR; } mEnabled = false; if (mEffect->suspended()) { return NO_ERROR; } status_t status = mEffect->setEnabled(false); sp thread = mEffect->thread().promote(); if (thread != 0) { thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); } return status; } void AudioFlinger::EffectHandle::disconnect() { disconnect(true); } void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) { ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); if (mEffect == 0) { return; } // restore suspended effects if the disconnected handle was enabled and the last one. if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { sp thread = mEffect->thread().promote(); if (thread != 0) { thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); } } // release sp on module => module destructor can be called now mEffect.clear(); if (mClient != 0) { if (mCblk != NULL) { // unlike ~TrackBase(), mCblk is never a local new, so don't delete mCblk->~effect_param_cblk_t(); // destroy our shared-structure. } mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to // Client destructor must run with AudioFlinger mutex locked Mutex::Autolock _l(mClient->audioFlinger()->mLock); mClient.clear(); } } status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, uint32_t cmdSize, void *pCmdData, uint32_t *replySize, void *pReplyData) { // ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", // cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); // only get parameter command is permitted for applications not controlling the effect if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { return INVALID_OPERATION; } if (mEffect == 0) return DEAD_OBJECT; if (mClient == 0) return INVALID_OPERATION; // handle commands that are not forwarded transparently to effect engine if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { // No need to trylock() here as this function is executed in the binder thread serving a particular client process: // no risk to block the whole media server process or mixer threads is we are stuck here Mutex::Autolock _l(mCblk->lock); if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { mCblk->serverIndex = 0; mCblk->clientIndex = 0; return BAD_VALUE; } status_t status = NO_ERROR; while (mCblk->serverIndex < mCblk->clientIndex) { int reply; uint32_t rsize = sizeof(int); int *p = (int *)(mBuffer + mCblk->serverIndex); int size = *p++; if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { ALOGW("command(): invalid parameter block size"); break; } effect_param_t *param = (effect_param_t *)p; if (param->psize == 0 || param->vsize == 0) { ALOGW("command(): null parameter or value size"); mCblk->serverIndex += size; continue; } uint32_t psize = sizeof(effect_param_t) + ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + param->vsize; status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, psize, p, &rsize, &reply); // stop at first error encountered if (ret != NO_ERROR) { status = ret; *(int *)pReplyData = reply; break; } else if (reply != NO_ERROR) { *(int *)pReplyData = reply; break; } mCblk->serverIndex += size; } mCblk->serverIndex = 0; mCblk->clientIndex = 0; return status; } else if (cmdCode == EFFECT_CMD_ENABLE) { *(int *)pReplyData = NO_ERROR; return enable(); } else if (cmdCode == EFFECT_CMD_DISABLE) { *(int *)pReplyData = NO_ERROR; return disable(); } #ifdef QCOM_HARDWARE ALOGV("EffectHandle::command: isOnLPA %d", mEffect->isOnLPA()); if(mEffect->isOnLPA() && ((cmdCode == EFFECT_CMD_SET_PARAM) || (cmdCode == EFFECT_CMD_SET_PARAM_DEFERRED) || (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) || (cmdCode == EFFECT_CMD_SET_DEVICE) || (cmdCode == EFFECT_CMD_SET_VOLUME) || (cmdCode == EFFECT_CMD_SET_AUDIO_MODE)) ) { // Notify Direct track for the change in Effect module // TODO: check if it is required to send mLPAHandle ALOGV("Notifying Direct Track for the change in effect config %d", cmdCode); mClient->audioFlinger()->audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL); } #endif return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); } void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) { ALOGV("setControl %p control %d", this, hasControl); mHasControl = hasControl; mEnabled = enabled; if (signal && mEffectClient != 0) { mEffectClient->controlStatusChanged(hasControl); } } void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, uint32_t cmdSize, void *pCmdData, uint32_t replySize, void *pReplyData) { if (mEffectClient != 0) { mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); } } void AudioFlinger::EffectHandle::setEnabled(bool enabled) { if (mEffectClient != 0) { mEffectClient->enableStatusChanged(enabled); } } status_t AudioFlinger::EffectHandle::onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { return BnEffect::onTransact(code, data, reply, flags); } void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) { bool locked = mCblk != NULL && tryLock(mCblk->lock); snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", (mClient == 0) ? getpid_cached : mClient->pid(), mPriority, mHasControl, !locked, mCblk ? mCblk->clientIndex : 0, mCblk ? mCblk->serverIndex : 0 ); if (locked) { mCblk->lock.unlock(); } } #undef LOG_TAG #define LOG_TAG "AudioFlinger::EffectChain" AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, int sessionId) : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) #ifdef QCOM_HARDWARE ,mIsForLPATrack(false) #endif { mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); if (thread == NULL) { return; } mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / thread->frameCount(); } AudioFlinger::EffectChain::~EffectChain() { if (mOwnInBuffer) { delete mInBuffer; } } // getEffectFromDesc_l() must be called with ThreadBase::mLock held sp AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) { size_t size = mEffects.size(); for (size_t i = 0; i < size; i++) { if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { return mEffects[i]; } } return 0; } // getEffectFromId_l() must be called with ThreadBase::mLock held sp AudioFlinger::EffectChain::getEffectFromId_l(int id) { size_t size = mEffects.size(); for (size_t i = 0; i < size; i++) { // by convention, return first effect if id provided is 0 (0 is never a valid id) if (id == 0 || mEffects[i]->id() == id) { return mEffects[i]; } } return 0; } #ifdef QCOM_HARDWARE sp AudioFlinger::EffectChain::getEffectFromIndex_l(int idx) { sp effect = NULL; if(idx < 0 || idx >= mEffects.size()) { ALOGE("EffectChain::getEffectFromIndex_l: invalid index %d", idx); } if(mEffects.size() > 0){ effect = mEffects[idx]; } return effect; } #endif // getEffectFromType_l() must be called with ThreadBase::mLock held sp AudioFlinger::EffectChain::getEffectFromType_l( const effect_uuid_t *type) { size_t size = mEffects.size(); for (size_t i = 0; i < size; i++) { if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { return mEffects[i]; } } return 0; } void AudioFlinger::EffectChain::clearInputBuffer() { Mutex::Autolock _l(mLock); sp thread = mThread.promote(); if (thread == 0) { ALOGW("clearInputBuffer(): cannot promote mixer thread"); return; } clearInputBuffer_l(thread); } // Must be called with EffectChain::mLock locked void AudioFlinger::EffectChain::clearInputBuffer_l(sp thread) { size_t numSamples = thread->frameCount() * thread->channelCount(); memset(mInBuffer, 0, numSamples * sizeof(int16_t)); } // Must be called with EffectChain::mLock locked void AudioFlinger::EffectChain::process_l() { sp thread = mThread.promote(); if (thread == 0) { ALOGW("process_l(): cannot promote mixer thread"); return; } bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); // always process effects unless no more tracks are on the session and the effect tail // has been rendered bool doProcess = true; if (!isGlobalSession) { bool tracksOnSession = (trackCnt() != 0); if (!tracksOnSession && mTailBufferCount == 0) { doProcess = false; } if (activeTrackCnt() == 0) { // if no track is active and the effect tail has not been rendered, // the input buffer must be cleared here as the mixer process will not do it if (tracksOnSession || mTailBufferCount > 0) { clearInputBuffer_l(thread); if (mTailBufferCount > 0) { mTailBufferCount--; } } } } size_t size = mEffects.size(); #ifdef QCOM_HARDWARE if (doProcess || isForLPATrack()) { #else if (doProcess) { #endif for (size_t i = 0; i < size; i++) { mEffects[i]->process(); } } for (size_t i = 0; i < size; i++) { mEffects[i]->updateState(); } } // addEffect_l() must be called with PlaybackThread::mLock held status_t AudioFlinger::EffectChain::addEffect_l(const sp& effect) { effect_descriptor_t desc = effect->desc(); uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; Mutex::Autolock _l(mLock); effect->setChain(this); sp thread = mThread.promote(); if (thread == 0) { return NO_INIT; } effect->setThread(thread); if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { // Auxiliary effects are inserted at the beginning of mEffects vector as // they are processed first and accumulated in chain input buffer mEffects.insertAt(effect, 0); // the input buffer for auxiliary effect contains mono samples in // 32 bit format. This is to avoid saturation in AudoMixer // accumulation stage. Saturation is done in EffectModule::process() before // calling the process in effect engine size_t numSamples = thread->frameCount(); int32_t *buffer = new int32_t[numSamples]; memset(buffer, 0, numSamples * sizeof(int32_t)); effect->setInBuffer((int16_t *)buffer); // auxiliary effects output samples to chain input buffer for further processing // by insert effects effect->setOutBuffer(mInBuffer); } else { // Insert effects are inserted at the end of mEffects vector as they are processed // after track and auxiliary effects. // Insert effect order as a function of indicated preference: // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if // another effect is present // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the // last effect claiming first position // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the // first effect claiming last position // else if EFFECT_FLAG_INSERT_ANY insert after first or before last // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is // already present size_t size = mEffects.size(); size_t idx_insert = size; ssize_t idx_insert_first = -1; ssize_t idx_insert_last = -1; for (size_t i = 0; i < size; i++) { effect_descriptor_t d = mEffects[i]->desc(); uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; if (iMode == EFFECT_FLAG_TYPE_INSERT) { // check invalid effect chaining combinations if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); return INVALID_OPERATION; } // remember position of first insert effect and by default // select this as insert position for new effect if (idx_insert == size) { idx_insert = i; } // remember position of last insert effect claiming // first position if (iPref == EFFECT_FLAG_INSERT_FIRST) { idx_insert_first = i; } // remember position of first insert effect claiming // last position if (iPref == EFFECT_FLAG_INSERT_LAST && idx_insert_last == -1) { idx_insert_last = i; } } } // modify idx_insert from first position if needed if (insertPref == EFFECT_FLAG_INSERT_LAST) { if (idx_insert_last != -1) { idx_insert = idx_insert_last; } else { idx_insert = size; } } else { if (idx_insert_first != -1) { idx_insert = idx_insert_first + 1; } } // always read samples from chain input buffer effect->setInBuffer(mInBuffer); // if last effect in the chain, output samples to chain // output buffer, otherwise to chain input buffer if (idx_insert == size) { if (idx_insert != 0) { mEffects[idx_insert-1]->setOutBuffer(mInBuffer); mEffects[idx_insert-1]->configure(); } effect->setOutBuffer(mOutBuffer); } else { effect->setOutBuffer(mInBuffer); } mEffects.insertAt(effect, idx_insert); ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); } effect->configure(); return NO_ERROR; } // removeEffect_l() must be called with PlaybackThread::mLock held size_t AudioFlinger::EffectChain::removeEffect_l(const sp& effect) { Mutex::Autolock _l(mLock); size_t size = mEffects.size(); uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; for (size_t i = 0; i < size; i++) { if (effect == mEffects[i]) { // calling stop here will remove pre-processing effect from the audio HAL. // This is safe as we hold the EffectChain mutex which guarantees that we are not in // the middle of a read from audio HAL if (mEffects[i]->state() == EffectModule::ACTIVE || mEffects[i]->state() == EffectModule::STOPPING) { mEffects[i]->stop(); } if (type == EFFECT_FLAG_TYPE_AUXILIARY) { delete[] effect->inBuffer(); } else { if (i == size - 1 && i != 0) { mEffects[i - 1]->setOutBuffer(mOutBuffer); mEffects[i - 1]->configure(); } } mEffects.removeAt(i); ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); break; } } return mEffects.size(); } // setDevice_l() must be called with PlaybackThread::mLock held void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) { size_t size = mEffects.size(); for (size_t i = 0; i < size; i++) { mEffects[i]->setDevice(device); } } // setMode_l() must be called with PlaybackThread::mLock held void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) { size_t size = mEffects.size(); for (size_t i = 0; i < size; i++) { mEffects[i]->setMode(mode); } } // setAudioSource_l() must be called with PlaybackThread::mLock held void AudioFlinger::EffectChain::setAudioSource_l(audio_source_t source) { size_t size = mEffects.size(); for (size_t i = 0; i < size; i++) { mEffects[i]->setAudioSource(source); } } // setVolume_l() must be called with PlaybackThread::mLock held bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) { uint32_t newLeft = *left; uint32_t newRight = *right; bool hasControl = false; int ctrlIdx = -1; size_t size = mEffects.size(); // first update volume controller for (size_t i = size; i > 0; i--) { if (mEffects[i - 1]->isProcessEnabled() && (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { ctrlIdx = i - 1; hasControl = true; break; } } if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { if (hasControl) { *left = mNewLeftVolume; *right = mNewRightVolume; } return hasControl; } mVolumeCtrlIdx = ctrlIdx; mLeftVolume = newLeft; mRightVolume = newRight; // second get volume update from volume controller if (ctrlIdx >= 0) { mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); mNewLeftVolume = newLeft; mNewRightVolume = newRight; } // then indicate volume to all other effects in chain. // Pass altered volume to effects before volume controller // and requested volume to effects after controller uint32_t lVol = newLeft; uint32_t rVol = newRight; for (size_t i = 0; i < size; i++) { if ((int)i == ctrlIdx) continue; // this also works for ctrlIdx == -1 when there is no volume controller if ((int)i > ctrlIdx) { lVol = *left; rVol = *right; } mEffects[i]->setVolume(&lVol, &rVol, false); } *left = newLeft; *right = newRight; return hasControl; } void AudioFlinger::EffectChain::dump(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); result.append(buffer); bool locked = tryLock(mLock); // failed to lock - AudioFlinger is probably deadlocked if (!locked) { result.append("\tCould not lock mutex:\n"); } result.append("\tNum fx In buffer Out buffer Active tracks:\n"); snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", mEffects.size(), (uint32_t)mInBuffer, (uint32_t)mOutBuffer, mActiveTrackCnt); result.append(buffer); write(fd, result.string(), result.size()); for (size_t i = 0; i < mEffects.size(); ++i) { sp effect = mEffects[i]; if (effect != 0) { effect->dump(fd, args); } } if (locked) { mLock.unlock(); } } // must be called with ThreadBase::mLock held void AudioFlinger::EffectChain::setEffectSuspended_l( const effect_uuid_t *type, bool suspend) { sp desc; // use effect type UUID timelow as key as there is no real risk of identical // timeLow fields among effect type UUIDs. ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); if (suspend) { if (index >= 0) { desc = mSuspendedEffects.valueAt(index); } else { desc = new SuspendedEffectDesc(); desc->mType = *type; mSuspendedEffects.add(type->timeLow, desc); ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); } if (desc->mRefCount++ == 0) { sp effect = getEffectIfEnabled(type); if (effect != 0) { desc->mEffect = effect; effect->setSuspended(true); effect->setEnabled(false); } } } else { if (index < 0) { return; } desc = mSuspendedEffects.valueAt(index); if (desc->mRefCount <= 0) { ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); desc->mRefCount = 1; } if (--desc->mRefCount == 0) { ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); if (desc->mEffect != 0) { sp effect = desc->mEffect.promote(); if (effect != 0) { effect->setSuspended(false); effect->lock(); EffectHandle *handle = effect->controlHandle_l(); if (handle != NULL && !handle->destroyed_l()) { effect->setEnabled_l(handle->enabled()); } effect->unlock(); } desc->mEffect.clear(); } mSuspendedEffects.removeItemsAt(index); } } } // must be called with ThreadBase::mLock held void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) { sp desc; ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); if (suspend) { if (index >= 0) { desc = mSuspendedEffects.valueAt(index); } else { desc = new SuspendedEffectDesc(); mSuspendedEffects.add((int)kKeyForSuspendAll, desc); ALOGV("setEffectSuspendedAll_l() add entry for 0"); } if (desc->mRefCount++ == 0) { Vector< sp > effects; getSuspendEligibleEffects(effects); for (size_t i = 0; i < effects.size(); i++) { setEffectSuspended_l(&effects[i]->desc().type, true); } } } else { if (index < 0) { return; } desc = mSuspendedEffects.valueAt(index); if (desc->mRefCount <= 0) { ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); desc->mRefCount = 1; } if (--desc->mRefCount == 0) { Vector types; for (size_t i = 0; i < mSuspendedEffects.size(); i++) { if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { continue; } types.add(&mSuspendedEffects.valueAt(i)->mType); } for (size_t i = 0; i < types.size(); i++) { setEffectSuspended_l(types[i], false); } ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); mSuspendedEffects.removeItem((int)kKeyForSuspendAll); } } } // The volume effect is used for automated tests only #ifndef OPENSL_ES_H_ static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; #endif //OPENSL_ES_H_ bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) { // auxiliary effects and visualizer are never suspended on output mix if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { return false; } return true; } void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp > &effects) { effects.clear(); for (size_t i = 0; i < mEffects.size(); i++) { if (isEffectEligibleForSuspend(mEffects[i]->desc())) { effects.add(mEffects[i]); } } } sp AudioFlinger::EffectChain::getEffectIfEnabled( const effect_uuid_t *type) { sp effect = getEffectFromType_l(type); return effect != 0 && effect->isEnabled() ? effect : 0; } void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp& effect, bool enabled) { ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); if (enabled) { if (index < 0) { // if the effect is not suspend check if all effects are suspended index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); if (index < 0) { return; } if (!isEffectEligibleForSuspend(effect->desc())) { return; } setEffectSuspended_l(&effect->desc().type, enabled); index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); if (index < 0) { ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); return; } } ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", effect->desc().type.timeLow); sp desc = mSuspendedEffects.valueAt(index); // if effect is requested to suspended but was not yet enabled, supend it now. if (desc->mEffect == 0) { desc->mEffect = effect; effect->setEnabled(false); effect->setSuspended(true); } } else { if (index < 0) { return; } ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", effect->desc().type.timeLow); sp desc = mSuspendedEffects.valueAt(index); desc->mEffect.clear(); effect->setSuspended(false); } } #undef LOG_TAG #define LOG_TAG "AudioFlinger" // ---------------------------------------------------------------------------- status_t AudioFlinger::onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { return BnAudioFlinger::onTransact(code, data, reply, flags); } }; // namespace android