| /* //device/include/server/AudioFlinger/AudioFlinger.cpp |
| ** |
| ** Copyright 2007, The Android Open Source Project |
| ** |
| ** Licensed under the Apache License, Version 2.0 (the "License"); |
| ** you may not use this file except in compliance with the License. |
| ** You may obtain a copy of the License at |
| ** |
| ** http://www.apache.org/licenses/LICENSE-2.0 |
| ** |
| ** Unless required by applicable law or agreed to in writing, software |
| ** distributed under the License is distributed on an "AS IS" BASIS, |
| ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| ** See the License for the specific language governing permissions and |
| ** limitations under the License. |
| */ |
| |
| |
| #define LOG_TAG "AudioFlinger" |
| //#define LOG_NDEBUG 0 |
| |
| #include <math.h> |
| #include <signal.h> |
| #include <sys/time.h> |
| #include <sys/resource.h> |
| |
| #include <binder/IPCThreadState.h> |
| #include <binder/IServiceManager.h> |
| #include <utils/Log.h> |
| #include <binder/Parcel.h> |
| #include <binder/IPCThreadState.h> |
| #include <utils/String16.h> |
| #include <utils/threads.h> |
| #include <utils/Atomic.h> |
| |
| #include <cutils/bitops.h> |
| #include <cutils/properties.h> |
| #include <cutils/compiler.h> |
| |
| #include <media/AudioTrack.h> |
| #include <media/AudioRecord.h> |
| #include <media/IMediaPlayerService.h> |
| |
| #include <private/media/AudioTrackShared.h> |
| #include <private/media/AudioEffectShared.h> |
| |
| #include <system/audio.h> |
| #include <hardware/audio.h> |
| |
| #include "AudioMixer.h" |
| #include "AudioFlinger.h" |
| |
| #include <media/EffectsFactoryApi.h> |
| #include <audio_effects/effect_visualizer.h> |
| #include <audio_effects/effect_ns.h> |
| #include <audio_effects/effect_aec.h> |
| |
| #include <audio_utils/primitives.h> |
| |
| #include <cpustats/ThreadCpuUsage.h> |
| #include <powermanager/PowerManager.h> |
| // #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds |
| |
| // ---------------------------------------------------------------------------- |
| |
| |
| namespace android { |
| |
| static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; |
| static const char kHardwareLockedString[] = "Hardware lock is taken\n"; |
| |
| //static const nsecs_t kStandbyTimeInNsecs = seconds(3); |
| static const float MAX_GAIN = 4096.0f; |
| static const float 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 = 2; |
| |
| 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; |
| |
| |
| // ---------------------------------------------------------------------------- |
| |
| static bool recordingAllowed() { |
| if (getpid() == IPCThreadState::self()->getCallingPid()) return true; |
| bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); |
| if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO"); |
| return ok; |
| } |
| |
| static bool settingsAllowed() { |
| if (getpid() == IPCThreadState::self()->getCallingPid()) return true; |
| bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); |
| if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); |
| return ok; |
| } |
| |
| // To collect the amplifier usage |
| static void addBatteryData(uint32_t params) { |
| sp<IBinder> binder = |
| defaultServiceManager()->getService(String16("media.player")); |
| sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder); |
| if (service.get() == NULL) { |
| LOGW("Cannot connect to the MediaPlayerService for battery tracking"); |
| return; |
| } |
| |
| service->addBatteryData(params); |
| } |
| |
| static int load_audio_interface(const char *if_name, const hw_module_t **mod, |
| audio_hw_device_t **dev) |
| { |
| int rc; |
| |
| rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); |
| if (rc) |
| goto out; |
| |
| rc = audio_hw_device_open(*mod, dev); |
| LOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", |
| AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); |
| if (rc) |
| goto out; |
| |
| return 0; |
| |
| out: |
| *mod = NULL; |
| *dev = NULL; |
| return rc; |
| } |
| |
| static const char * const audio_interfaces[] = { |
| "primary", |
| "a2dp", |
| "usb", |
| }; |
| #define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) |
| |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::AudioFlinger() |
| : BnAudioFlinger(), |
| mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), |
| mBtNrecIsOff(false) |
| { |
| } |
| |
| void AudioFlinger::onFirstRef() |
| { |
| int rc = 0; |
| |
| Mutex::Autolock _l(mLock); |
| |
| /* TODO: move all this work into an Init() function */ |
| mHardwareStatus = AUDIO_HW_IDLE; |
| |
| for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { |
| const hw_module_t *mod; |
| audio_hw_device_t *dev; |
| |
| rc = load_audio_interface(audio_interfaces[i], &mod, &dev); |
| if (rc) |
| continue; |
| |
| LOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], |
| mod->name, mod->id); |
| mAudioHwDevs.push(dev); |
| |
| if (!mPrimaryHardwareDev) { |
| mPrimaryHardwareDev = dev; |
| LOGI("Using '%s' (%s.%s) as the primary audio interface", |
| mod->name, mod->id, audio_interfaces[i]); |
| } |
| } |
| |
| mHardwareStatus = AUDIO_HW_INIT; |
| |
| if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) { |
| LOGE("Primary audio interface not found"); |
| return; |
| } |
| |
| for (size_t i = 0; i < mAudioHwDevs.size(); i++) { |
| audio_hw_device_t *dev = mAudioHwDevs[i]; |
| |
| mHardwareStatus = AUDIO_HW_INIT; |
| rc = dev->init_check(dev); |
| if (rc == 0) { |
| AutoMutex lock(mHardwareLock); |
| |
| mMode = AUDIO_MODE_NORMAL; |
| mHardwareStatus = AUDIO_HW_SET_MODE; |
| dev->set_mode(dev, mMode); |
| mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; |
| dev->set_master_volume(dev, 1.0f); |
| mHardwareStatus = AUDIO_HW_IDLE; |
| } |
| } |
| } |
| |
| status_t AudioFlinger::initCheck() const |
| { |
| Mutex::Autolock _l(mLock); |
| if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0) |
| return NO_INIT; |
| return NO_ERROR; |
| } |
| |
| AudioFlinger::~AudioFlinger() |
| { |
| int num_devs = mAudioHwDevs.size(); |
| |
| while (!mRecordThreads.isEmpty()) { |
| // closeInput() will remove first entry from mRecordThreads |
| closeInput(mRecordThreads.keyAt(0)); |
| } |
| while (!mPlaybackThreads.isEmpty()) { |
| // closeOutput() will remove first entry from mPlaybackThreads |
| closeOutput(mPlaybackThreads.keyAt(0)); |
| } |
| |
| for (int i = 0; i < num_devs; i++) { |
| audio_hw_device_t *dev = mAudioHwDevs[i]; |
| audio_hw_device_close(dev); |
| } |
| mAudioHwDevs.clear(); |
| } |
| |
| audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) |
| { |
| /* first matching HW device is returned */ |
| for (size_t i = 0; i < mAudioHwDevs.size(); i++) { |
| audio_hw_device_t *dev = mAudioHwDevs[i]; |
| if ((dev->get_supported_devices(dev) & devices) == devices) |
| return dev; |
| } |
| return NULL; |
| } |
| |
| status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) |
| { |
| const size_t SIZE = 256; |
| char buffer[SIZE]; |
| String8 result; |
| |
| result.append("Clients:\n"); |
| for (size_t i = 0; i < mClients.size(); ++i) { |
| wp<Client> wClient = mClients.valueAt(i); |
| if (wClient != 0) { |
| sp<Client> client = wClient.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 cnt\n"); |
| for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { |
| AudioSessionRef *r = mAudioSessionRefs[i]; |
| snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); |
| result.append(buffer); |
| } |
| write(fd, result.string(), result.size()); |
| return NO_ERROR; |
| } |
| |
| |
| status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) |
| { |
| const size_t SIZE = 256; |
| char buffer[SIZE]; |
| String8 result; |
| int hardwareStatus = mHardwareStatus; |
| |
| snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); |
| result.append(buffer); |
| write(fd, result.string(), result.size()); |
| return NO_ERROR; |
| } |
| |
| status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& 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()); |
| return NO_ERROR; |
| } |
| |
| 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<String16>& args) |
| { |
| if (checkCallingPermission(String16("android.permission.DUMP")) == false) { |
| 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[i]; |
| dev->dump(dev, fd); |
| } |
| if (locked) mLock.unlock(); |
| } |
| return NO_ERROR; |
| } |
| |
| |
| // IAudioFlinger interface |
| |
| |
| sp<IAudioTrack> AudioFlinger::createTrack( |
| pid_t pid, |
| int streamType, |
| uint32_t sampleRate, |
| uint32_t format, |
| uint32_t channelMask, |
| int frameCount, |
| uint32_t flags, |
| const sp<IMemory>& sharedBuffer, |
| int output, |
| int *sessionId, |
| status_t *status) |
| { |
| sp<PlaybackThread::Track> track; |
| sp<TrackHandle> trackHandle; |
| sp<Client> client; |
| wp<Client> wclient; |
| status_t lStatus; |
| int lSessionId; |
| |
| if (streamType >= AUDIO_STREAM_CNT) { |
| LOGE("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) { |
| LOGE("unknown output thread"); |
| lStatus = BAD_VALUE; |
| goto Exit; |
| } |
| |
| wclient = mClients.valueFor(pid); |
| |
| if (wclient != NULL) { |
| client = wclient.promote(); |
| } else { |
| client = new Client(this, pid); |
| mClients.add(pid, client); |
| } |
| |
| ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); |
| if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { |
| for (size_t i = 0; i < mPlaybackThreads.size(); i++) { |
| sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); |
| if (mPlaybackThreads.keyAt(i) != output) { |
| // prevent same audio session on different output threads |
| uint32_t sessions = t->hasAudioSession(*sessionId); |
| if (sessions & PlaybackThread::TRACK_SESSION) { |
| LOGE("createTrack() session ID %d already in use", *sessionId); |
| lStatus = BAD_VALUE; |
| goto Exit; |
| } |
| // check if an effect with same session ID is waiting for a track to be created |
| if (sessions & PlaybackThread::EFFECT_SESSION) { |
| effectThread = t.get(); |
| } |
| } |
| } |
| 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, &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); |
| } |
| } |
| 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) { |
| *status = lStatus; |
| } |
| return trackHandle; |
| } |
| |
| uint32_t AudioFlinger::sampleRate(int output) const |
| { |
| Mutex::Autolock _l(mLock); |
| PlaybackThread *thread = checkPlaybackThread_l(output); |
| if (thread == NULL) { |
| LOGW("sampleRate() unknown thread %d", output); |
| return 0; |
| } |
| return thread->sampleRate(); |
| } |
| |
| int AudioFlinger::channelCount(int output) const |
| { |
| Mutex::Autolock _l(mLock); |
| PlaybackThread *thread = checkPlaybackThread_l(output); |
| if (thread == NULL) { |
| LOGW("channelCount() unknown thread %d", output); |
| return 0; |
| } |
| return thread->channelCount(); |
| } |
| |
| uint32_t AudioFlinger::format(int output) const |
| { |
| Mutex::Autolock _l(mLock); |
| PlaybackThread *thread = checkPlaybackThread_l(output); |
| if (thread == NULL) { |
| LOGW("format() unknown thread %d", output); |
| return 0; |
| } |
| return thread->format(); |
| } |
| |
| size_t AudioFlinger::frameCount(int output) const |
| { |
| Mutex::Autolock _l(mLock); |
| PlaybackThread *thread = checkPlaybackThread_l(output); |
| if (thread == NULL) { |
| LOGW("frameCount() unknown thread %d", output); |
| return 0; |
| } |
| return thread->frameCount(); |
| } |
| |
| uint32_t AudioFlinger::latency(int output) const |
| { |
| Mutex::Autolock _l(mLock); |
| PlaybackThread *thread = checkPlaybackThread_l(output); |
| if (thread == NULL) { |
| LOGW("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; |
| } |
| |
| // when hw supports master volume, don't scale in sw mixer |
| { // scope for the lock |
| AutoMutex lock(mHardwareLock); |
| mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; |
| if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { |
| value = 1.0f; |
| } |
| mHardwareStatus = AUDIO_HW_IDLE; |
| } |
| |
| Mutex::Autolock _l(mLock); |
| mMasterVolume = value; |
| for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) |
| mPlaybackThreads.valueAt(i)->setMasterVolume(value); |
| |
| return NO_ERROR; |
| } |
| |
| status_t AudioFlinger::setMode(int mode) |
| { |
| status_t ret = initCheck(); |
| if (ret != NO_ERROR) { |
| return ret; |
| } |
| |
| // check calling permissions |
| if (!settingsAllowed()) { |
| return PERMISSION_DENIED; |
| } |
| if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) { |
| LOGW("Illegal value: setMode(%d)", mode); |
| return BAD_VALUE; |
| } |
| |
| { // scope for the lock |
| AutoMutex lock(mHardwareLock); |
| mHardwareStatus = AUDIO_HW_SET_MODE; |
| ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); |
| mHardwareStatus = AUDIO_HW_IDLE; |
| } |
| |
| if (NO_ERROR == ret) { |
| Mutex::Autolock _l(mLock); |
| mMode = mode; |
| for (uint32_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); |
| mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; |
| ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, 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; |
| mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; |
| mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); |
| mHardwareStatus = AUDIO_HW_IDLE; |
| return state; |
| } |
| |
| status_t AudioFlinger::setMasterMute(bool muted) |
| { |
| // check calling permissions |
| if (!settingsAllowed()) { |
| return PERMISSION_DENIED; |
| } |
| |
| Mutex::Autolock _l(mLock); |
| mMasterMute = muted; |
| for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) |
| mPlaybackThreads.valueAt(i)->setMasterMute(muted); |
| |
| return NO_ERROR; |
| } |
| |
| float AudioFlinger::masterVolume() const |
| { |
| return mMasterVolume; |
| } |
| |
| bool AudioFlinger::masterMute() const |
| { |
| return mMasterMute; |
| } |
| |
| status_t AudioFlinger::setStreamVolume(int stream, float value, int output) |
| { |
| // check calling permissions |
| if (!settingsAllowed()) { |
| return PERMISSION_DENIED; |
| } |
| |
| if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { |
| LOGE("setStreamVolume() invalid stream %d", stream); |
| return BAD_VALUE; |
| } |
| |
| AutoMutex lock(mLock); |
| PlaybackThread *thread = NULL; |
| if (output) { |
| thread = checkPlaybackThread_l(output); |
| if (thread == NULL) { |
| return BAD_VALUE; |
| } |
| } |
| |
| mStreamTypes[stream].volume = value; |
| |
| if (thread == NULL) { |
| for (uint32_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(int stream, bool muted) |
| { |
| // check calling permissions |
| if (!settingsAllowed()) { |
| return PERMISSION_DENIED; |
| } |
| |
| if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT || |
| uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { |
| LOGE("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(int stream, int output) const |
| { |
| if (stream < 0 || 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 = mStreamTypes[stream].volume; |
| } |
| |
| return volume; |
| } |
| |
| bool AudioFlinger::streamMute(int stream) const |
| { |
| if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) { |
| return true; |
| } |
| |
| return mStreamTypes[stream].mute; |
| } |
| |
| status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) |
| { |
| status_t result; |
| |
| ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %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) { |
| AutoMutex lock(mHardwareLock); |
| mHardwareStatus = AUDIO_SET_PARAMETER; |
| status_t final_result = NO_ERROR; |
| for (size_t i = 0; i < mAudioHwDevs.size(); i++) { |
| audio_hw_device_t *dev = mAudioHwDevs[i]; |
| result = dev->set_parameters(dev, keyValuePairs.string()); |
| final_result = result ?: final_result; |
| } |
| mHardwareStatus = AUDIO_HW_IDLE; |
| // disable AEC and NS if the device is a BT SCO headset supporting those pre processings |
| AudioParameter param = AudioParameter(keyValuePairs); |
| String8 value; |
| if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { |
| Mutex::Autolock _l(mLock); |
| bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); |
| if (mBtNrecIsOff != btNrecIsOff) { |
| for (size_t i = 0; i < mRecordThreads.size(); i++) { |
| sp<RecordThread> thread = mRecordThreads.valueAt(i); |
| RecordThread::RecordTrack *track = thread->track(); |
| if (track != NULL) { |
| audio_devices_t device = (audio_devices_t)( |
| thread->device() & AUDIO_DEVICE_IN_ALL); |
| bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; |
| thread->setEffectSuspended(FX_IID_AEC, |
| suspend, |
| track->sessionId()); |
| thread->setEffectSuspended(FX_IID_NS, |
| suspend, |
| track->sessionId()); |
| } |
| } |
| mBtNrecIsOff = btNrecIsOff; |
| } |
| } |
| return final_result; |
| } |
| |
| // hold a strong ref on thread in case closeOutput() or closeInput() is called |
| // and the thread is exited once the lock is released |
| sp<ThreadBase> thread; |
| { |
| Mutex::Autolock _l(mLock); |
| thread = checkPlaybackThread_l(ioHandle); |
| if (thread == NULL) { |
| thread = checkRecordThread_l(ioHandle); |
| } else if (thread.get() == primaryPlaybackThread_l()) { |
| // indicate output device change to all input threads for pre processing |
| AudioParameter param = AudioParameter(keyValuePairs); |
| int value; |
| if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { |
| for (size_t i = 0; i < mRecordThreads.size(); i++) { |
| mRecordThreads.valueAt(i)->setParameters(keyValuePairs); |
| } |
| } |
| } |
| } |
| if (thread != NULL) { |
| result = thread->setParameters(keyValuePairs); |
| return result; |
| } |
| return BAD_VALUE; |
| } |
| |
| String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) |
| { |
| // ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", |
| // ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); |
| |
| if (ioHandle == 0) { |
| String8 out_s8; |
| |
| for (size_t i = 0; i < mAudioHwDevs.size(); i++) { |
| audio_hw_device_t *dev = mAudioHwDevs[i]; |
| char *s = dev->get_parameters(dev, keys.string()); |
| out_s8 += String8(s); |
| free(s); |
| } |
| return out_s8; |
| } |
| |
| Mutex::Autolock _l(mLock); |
| |
| 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, int format, int channelCount) |
| { |
| status_t ret = initCheck(); |
| if (ret != NO_ERROR) { |
| return 0; |
| } |
| |
| return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); |
| } |
| |
| unsigned int AudioFlinger::getInputFramesLost(int ioHandle) |
| { |
| if (ioHandle == 0) { |
| return 0; |
| } |
| |
| 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); |
| mHardwareStatus = AUDIO_SET_VOICE_VOLUME; |
| ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); |
| mHardwareStatus = AUDIO_HW_IDLE; |
| |
| return ret; |
| } |
| |
| status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) |
| { |
| status_t status; |
| |
| Mutex::Autolock _l(mLock); |
| |
| PlaybackThread *playbackThread = checkPlaybackThread_l(output); |
| if (playbackThread != NULL) { |
| return playbackThread->getRenderPosition(halFrames, dspFrames); |
| } |
| |
| return BAD_VALUE; |
| } |
| |
| void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) |
| { |
| |
| Mutex::Autolock _l(mLock); |
| |
| int pid = IPCThreadState::self()->getCallingPid(); |
| if (mNotificationClients.indexOfKey(pid) < 0) { |
| sp<NotificationClient> notificationClient = new NotificationClient(this, |
| client, |
| pid); |
| ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); |
| |
| mNotificationClients.add(pid, notificationClient); |
| |
| sp<IBinder> 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)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); |
| } |
| |
| for (size_t i = 0; i < mRecordThreads.size(); i++) { |
| mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); |
| } |
| } |
| } |
| |
| void AudioFlinger::removeNotificationClient(pid_t pid) |
| { |
| Mutex::Autolock _l(mLock); |
| |
| int index = mNotificationClients.indexOfKey(pid); |
| if (index >= 0) { |
| sp <NotificationClient> client = mNotificationClients.valueFor(pid); |
| ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); |
| mNotificationClients.removeItem(pid); |
| } |
| |
| ALOGV("%d died, releasing its sessions", pid); |
| int num = mAudioSessionRefs.size(); |
| bool removed = false; |
| for (int i = 0; i< num; i++) { |
| AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); |
| ALOGV(" pid %d @ %d", ref->pid, i); |
| if (ref->pid == pid) { |
| ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); |
| mAudioSessionRefs.removeAt(i); |
| delete ref; |
| removed = true; |
| i--; |
| num--; |
| } |
| } |
| if (removed) { |
| purgeStaleEffects_l(); |
| } |
| } |
| |
| // audioConfigChanged_l() must be called with AudioFlinger::mLock held |
| void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) |
| { |
| size_t size = mNotificationClients.size(); |
| for (size_t i = 0; i < size; i++) { |
| mNotificationClients.valueAt(i)->client()->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); |
| } |
| |
| |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device) |
| : Thread(false), |
| mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), |
| mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false), |
| mDevice(device) |
| { |
| mDeathRecipient = new PMDeathRecipient(this); |
| } |
| |
| AudioFlinger::ThreadBase::~ThreadBase() |
| { |
| mParamCond.broadcast(); |
| // do not lock the mutex in destructor |
| releaseWakeLock_l(); |
| if (mPowerManager != 0) { |
| sp<IBinder> binder = mPowerManager->asBinder(); |
| binder->unlinkToDeath(mDeathRecipient); |
| } |
| } |
| |
| void AudioFlinger::ThreadBase::exit() |
| { |
| // keep a strong ref on ourself so that we won't get |
| // destroyed in the middle of requestExitAndWait() |
| sp <ThreadBase> strongMe = this; |
| |
| ALOGV("ThreadBase::exit"); |
| { |
| AutoMutex lock(&mLock); |
| mExiting = true; |
| requestExit(); |
| mWaitWorkCV.signal(); |
| } |
| requestExitAndWait(); |
| } |
| |
| uint32_t AudioFlinger::ThreadBase::sampleRate() const |
| { |
| return mSampleRate; |
| } |
| |
| int AudioFlinger::ThreadBase::channelCount() const |
| { |
| return (int)mChannelCount; |
| } |
| |
| uint32_t AudioFlinger::ThreadBase::format() const |
| { |
| return mFormat; |
| } |
| |
| size_t AudioFlinger::ThreadBase::frameCount() const |
| { |
| return mFrameCount; |
| } |
| |
| 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; |
| } |
| |
| void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) |
| { |
| Mutex::Autolock _l(mLock); |
| sendConfigEvent_l(event, param); |
| } |
| |
| // sendConfigEvent_l() must be called with ThreadBase::mLock held |
| void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) |
| { |
| ConfigEvent configEvent; |
| configEvent.mEvent = event; |
| configEvent.mParam = param; |
| mConfigEvents.add(configEvent); |
| ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); |
| mWaitWorkCV.signal(); |
| } |
| |
| void AudioFlinger::ThreadBase::processConfigEvents() |
| { |
| mLock.lock(); |
| while(!mConfigEvents.isEmpty()) { |
| ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); |
| ConfigEvent configEvent = mConfigEvents[0]; |
| mConfigEvents.removeAt(0); |
| // release mLock before locking AudioFlinger mLock: lock order is always |
| // AudioFlinger then ThreadBase to avoid cross deadlock |
| mLock.unlock(); |
| mAudioFlinger->mLock.lock(); |
| audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); |
| mAudioFlinger->mLock.unlock(); |
| mLock.lock(); |
| } |
| mLock.unlock(); |
| } |
| |
| status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& 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, "standby: %d\n", mStandby); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); |
| 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: %d\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); |
| snprintf(buffer, SIZE, " Index event param\n"); |
| result.append(buffer); |
| for (size_t i = 0; i < mConfigEvents.size(); i++) { |
| snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); |
| result.append(buffer); |
| } |
| result.append("\n"); |
| |
| write(fd, result.string(), result.size()); |
| |
| if (locked) { |
| mLock.unlock(); |
| } |
| return NO_ERROR; |
| } |
| |
| status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& 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<EffectChain> chain = mEffectChains[i]; |
| if (chain != 0) { |
| chain->dump(fd, args); |
| } |
| } |
| return NO_ERROR; |
| } |
| |
| 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<IBinder> binder = |
| defaultServiceManager()->checkService(String16("power")); |
| if (binder == 0) { |
| LOGW("Thread %s cannot connect to the power manager service", mName); |
| } else { |
| mPowerManager = interface_cast<IPowerManager>(binder); |
| binder->linkToDeath(mDeathRecipient); |
| } |
| } |
| if (mPowerManager != 0) { |
| sp<IBinder> 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<IBinder>& who) |
| { |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| thread->clearPowerManager(); |
| } |
| LOGW("power manager service died !!!"); |
| } |
| |
| void AudioFlinger::ThreadBase::setEffectSuspended( |
| const effect_uuid_t *type, bool suspend, int sessionId) |
| { |
| Mutex::Autolock _l(mLock); |
| setEffectSuspended_l(type, suspend, sessionId); |
| } |
| |
| void AudioFlinger::ThreadBase::setEffectSuspended_l( |
| const effect_uuid_t *type, bool suspend, int sessionId) |
| { |
| sp<EffectChain> chain; |
| chain = getEffectChain_l(sessionId); |
| if (chain != 0) { |
| if (type != NULL) { |
| chain->setEffectSuspended_l(type, suspend); |
| } else { |
| chain->setEffectSuspendedAll_l(suspend); |
| } |
| } |
| |
| updateSuspendedSessions_l(type, suspend, sessionId); |
| } |
| |
| void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) |
| { |
| int index = mSuspendedSessions.indexOfKey(chain->sessionId()); |
| if (index < 0) { |
| return; |
| } |
| |
| KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = |
| mSuspendedSessions.editValueAt(index); |
| |
| for (size_t i = 0; i < sessionEffects.size(); i++) { |
| sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); |
| for (int j = 0; j < desc->mRefCount; j++) { |
| if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { |
| chain->setEffectSuspendedAll_l(true); |
| } else { |
| ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", |
| desc->mType.timeLow); |
| chain->setEffectSuspended_l(&desc->mType, true); |
| } |
| } |
| } |
| } |
| |
| void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, |
| bool suspend, |
| int sessionId) |
| { |
| int index = mSuspendedSessions.indexOfKey(sessionId); |
| |
| KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; |
| |
| if (suspend) { |
| if (index >= 0) { |
| sessionEffects = mSuspendedSessions.editValueAt(index); |
| } else { |
| mSuspendedSessions.add(sessionId, sessionEffects); |
| } |
| } else { |
| if (index < 0) { |
| return; |
| } |
| sessionEffects = mSuspendedSessions.editValueAt(index); |
| } |
| |
| |
| int key = EffectChain::kKeyForSuspendAll; |
| if (type != NULL) { |
| key = type->timeLow; |
| } |
| index = sessionEffects.indexOfKey(key); |
| |
| sp <SuspendedSessionDesc> desc; |
| if (suspend) { |
| if (index >= 0) { |
| desc = sessionEffects.valueAt(index); |
| } else { |
| desc = new SuspendedSessionDesc(); |
| if (type != NULL) { |
| memcpy(&desc->mType, type, sizeof(effect_uuid_t)); |
| } |
| sessionEffects.add(key, desc); |
| ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); |
| } |
| desc->mRefCount++; |
| } else { |
| if (index < 0) { |
| return; |
| } |
| desc = sessionEffects.valueAt(index); |
| if (--desc->mRefCount == 0) { |
| ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); |
| sessionEffects.removeItemsAt(index); |
| if (sessionEffects.isEmpty()) { |
| ALOGV("updateSuspendedSessions_l() restore removing session %d", |
| sessionId); |
| mSuspendedSessions.removeItem(sessionId); |
| } |
| } |
| } |
| if (!sessionEffects.isEmpty()) { |
| mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); |
| } |
| } |
| |
| void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, |
| bool enabled, |
| int sessionId) |
| { |
| Mutex::Autolock _l(mLock); |
| checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); |
| } |
| |
| void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, |
| bool enabled, |
| int sessionId) |
| { |
| if (mType != RECORD) { |
| // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on |
| // another session. This gives the priority to well behaved effect control panels |
| // and applications not using global effects. |
| if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { |
| setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); |
| } |
| } |
| |
| sp<EffectChain> chain = getEffectChain_l(sessionId); |
| if (chain != 0) { |
| chain->checkSuspendOnEffectEnabled(effect, enabled); |
| } |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, |
| AudioStreamOut* output, |
| int id, |
| uint32_t device) |
| : ThreadBase(audioFlinger, id, device), |
| mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output), |
| mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) |
| { |
| snprintf(mName, kNameLength, "AudioOut_%d", id); |
| |
| readOutputParameters(); |
| |
| mMasterVolume = mAudioFlinger->masterVolume(); |
| mMasterMute = mAudioFlinger->masterMute(); |
| |
| for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { |
| mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); |
| mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); |
| mStreamTypes[stream].valid = true; |
| } |
| } |
| |
| AudioFlinger::PlaybackThread::~PlaybackThread() |
| { |
| delete [] mMixBuffer; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) |
| { |
| dumpInternals(fd, args); |
| dumpTracks(fd, args); |
| dumpEffectChains(fd, args); |
| return NO_ERROR; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) |
| { |
| const size_t SIZE = 256; |
| char buffer[SIZE]; |
| String8 result; |
| |
| snprintf(buffer, SIZE, "Output thread %p tracks\n", this); |
| result.append(buffer); |
| result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); |
| for (size_t i = 0; i < mTracks.size(); ++i) { |
| sp<Track> 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); |
| result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); |
| for (size_t i = 0; i < mActiveTracks.size(); ++i) { |
| wp<Track> wTrack = mActiveTracks[i]; |
| if (wTrack != 0) { |
| sp<Track> track = wTrack.promote(); |
| if (track != 0) { |
| track->dump(buffer, SIZE); |
| result.append(buffer); |
| } |
| } |
| } |
| write(fd, result.string(), result.size()); |
| return NO_ERROR; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& 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()); |
| |
| dumpBase(fd, args); |
| |
| return NO_ERROR; |
| } |
| |
| // Thread virtuals |
| status_t AudioFlinger::PlaybackThread::readyToRun() |
| { |
| status_t status = initCheck(); |
| if (status == NO_ERROR) { |
| LOGI("AudioFlinger's thread %p ready to run", this); |
| } else { |
| LOGE("No working audio driver found."); |
| } |
| return status; |
| } |
| |
| void AudioFlinger::PlaybackThread::onFirstRef() |
| { |
| run(mName, ANDROID_PRIORITY_URGENT_AUDIO); |
| } |
| |
| // PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held |
| sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( |
| const sp<AudioFlinger::Client>& client, |
| int streamType, |
| uint32_t sampleRate, |
| uint32_t format, |
| uint32_t channelMask, |
| int frameCount, |
| const sp<IMemory>& sharedBuffer, |
| int sessionId, |
| status_t *status) |
| { |
| sp<Track> track; |
| status_t lStatus; |
| |
| if (mType == DIRECT) { |
| if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { |
| if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { |
| LOGE("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) { |
| LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); |
| lStatus = BAD_VALUE; |
| goto Exit; |
| } |
| } |
| |
| lStatus = initCheck(); |
| if (lStatus != NO_ERROR) { |
| LOGE("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((audio_stream_type_t)streamType); |
| for (size_t i = 0; i < mTracks.size(); ++i) { |
| sp<Track> t = mTracks[i]; |
| if (t != 0) { |
| uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type()); |
| if (sessionId == t->sessionId() && strategy != actual) { |
| LOGE("createTrack_l() mismatched strategy; expected %u but found %u", |
| strategy, actual); |
| lStatus = BAD_VALUE; |
| goto Exit; |
| } |
| } |
| } |
| |
| track = new Track(this, client, streamType, sampleRate, format, |
| channelMask, frameCount, sharedBuffer, sessionId); |
| if (track->getCblk() == NULL || track->name() < 0) { |
| lStatus = NO_MEMORY; |
| goto Exit; |
| } |
| mTracks.add(track); |
| |
| sp<EffectChain> chain = getEffectChain_l(sessionId); |
| if (chain != 0) { |
| ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); |
| track->setMainBuffer(chain->inBuffer()); |
| chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); |
| chain->incTrackCnt(); |
| } |
| |
| // invalidate track immediately if the stream type was moved to another thread since |
| // createTrack() was called by the client process. |
| if (!mStreamTypes[streamType].valid) { |
| LOGW("createTrack_l() on thread %p: invalidating track on stream %d", |
| this, streamType); |
| android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); |
| } |
| } |
| lStatus = NO_ERROR; |
| |
| Exit: |
| if(status) { |
| *status = lStatus; |
| } |
| return track; |
| } |
| |
| uint32_t AudioFlinger::PlaybackThread::latency() const |
| { |
| Mutex::Autolock _l(mLock); |
| if (initCheck() == NO_ERROR) { |
| return mOutput->stream->get_latency(mOutput->stream); |
| } else { |
| return 0; |
| } |
| } |
| |
| status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) |
| { |
| mMasterVolume = value; |
| return NO_ERROR; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) |
| { |
| mMasterMute = muted; |
| return NO_ERROR; |
| } |
| |
| float AudioFlinger::PlaybackThread::masterVolume() const |
| { |
| return mMasterVolume; |
| } |
| |
| bool AudioFlinger::PlaybackThread::masterMute() const |
| { |
| return mMasterMute; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) |
| { |
| mStreamTypes[stream].volume = value; |
| return NO_ERROR; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) |
| { |
| mStreamTypes[stream].mute = muted; |
| return NO_ERROR; |
| } |
| |
| float AudioFlinger::PlaybackThread::streamVolume(int stream) const |
| { |
| return mStreamTypes[stream].volume; |
| } |
| |
| bool AudioFlinger::PlaybackThread::streamMute(int stream) const |
| { |
| return mStreamTypes[stream].mute; |
| } |
| |
| // addTrack_l() must be called with ThreadBase::mLock held |
| status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) |
| { |
| status_t status = ALREADY_EXISTS; |
| |
| // set retry count for buffer fill |
| track->mRetryCount = kMaxTrackStartupRetries; |
| if (mActiveTracks.indexOf(track) < 0) { |
| // the track is newly added, make sure it fills up all its |
| // buffers before playing. This is to ensure the client will |
| // effectively get the latency it requested. |
| track->mFillingUpStatus = Track::FS_FILLING; |
| track->mResetDone = false; |
| mActiveTracks.add(track); |
| if (track->mainBuffer() != mMixBuffer) { |
| sp<EffectChain> chain = getEffectChain_l(track->sessionId()); |
| if (chain != 0) { |
| ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); |
| chain->incActiveTrackCnt(); |
| } |
| } |
| |
| status = NO_ERROR; |
| } |
| |
| 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) |
| { |
| track->mState = TrackBase::TERMINATED; |
| if (mActiveTracks.indexOf(track) < 0) { |
| removeTrack_l(track); |
| } |
| } |
| |
| void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) |
| { |
| mTracks.remove(track); |
| deleteTrackName_l(track->name()); |
| sp<EffectChain> 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 = 0; |
| |
| 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 = mFrameCount; |
| 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 = (uint16_t)audio_stream_frame_size(&mOutput->stream->common); |
| mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; |
| |
| // FIXME - Current mixer implementation only supports stereo output: Always |
| // Allocate a stereo buffer even if HW output is mono. |
| if (mMixBuffer != NULL) delete[] mMixBuffer; |
| mMixBuffer = new int16_t[mFrameCount * 2]; |
| memset(mMixBuffer, 0, mFrameCount * 2 * 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<EffectChain> > effectChains = mEffectChains; |
| for (size_t i = 0; i < effectChains.size(); i ++) { |
| mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); |
| } |
| } |
| |
| status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) |
| { |
| if (halFrames == 0 || dspFrames == 0) { |
| return BAD_VALUE; |
| } |
| Mutex::Autolock _l(mLock); |
| if (initCheck() != NO_ERROR) { |
| return INVALID_OPERATION; |
| } |
| *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); |
| |
| return mOutput->stream->get_render_position(mOutput->stream, dspFrames); |
| } |
| |
| uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) |
| { |
| Mutex::Autolock _l(mLock); |
| uint32_t result = 0; |
| if (getEffectChain_l(sessionId) != 0) { |
| result = EFFECT_SESSION; |
| } |
| |
| for (size_t i = 0; i < mTracks.size(); ++i) { |
| sp<Track> track = mTracks[i]; |
| if (sessionId == track->sessionId() && |
| !(track->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> track = mTracks[i]; |
| if (sessionId == track->sessionId() && |
| !(track->mCblk->flags & CBLK_INVALID_MSK)) { |
| return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); |
| } |
| } |
| return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); |
| } |
| |
| |
| AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() |
| { |
| Mutex::Autolock _l(mLock); |
| return mOutput; |
| } |
| |
| AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() |
| { |
| Mutex::Autolock _l(mLock); |
| AudioStreamOut *output = mOutput; |
| mOutput = NULL; |
| return output; |
| } |
| |
| // this method must always be called either with ThreadBase mLock held or inside the thread loop |
| audio_stream_t* AudioFlinger::PlaybackThread::stream() |
| { |
| if (mOutput == NULL) { |
| return NULL; |
| } |
| return &mOutput->stream->common; |
| } |
| |
| uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() |
| { |
| // A2DP output latency is not due only to buffering capacity. It also reflects encoding, |
| // decoding and transfer time. So sleeping for half of the latency would likely cause |
| // underruns |
| if (audio_is_a2dp_device((audio_devices_t)mDevice)) { |
| return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); |
| } else { |
| return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; |
| } |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) |
| : PlaybackThread(audioFlinger, output, id, device), |
| mAudioMixer(NULL) |
| { |
| mType = ThreadBase::MIXER; |
| mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); |
| |
| // FIXME - Current mixer implementation only supports stereo output |
| if (mChannelCount == 1) { |
| LOGE("Invalid audio hardware channel count"); |
| } |
| } |
| |
| AudioFlinger::MixerThread::~MixerThread() |
| { |
| delete mAudioMixer; |
| } |
| |
| bool AudioFlinger::MixerThread::threadLoop() |
| { |
| Vector< sp<Track> > tracksToRemove; |
| uint32_t mixerStatus = MIXER_IDLE; |
| nsecs_t standbyTime = systemTime(); |
| size_t mixBufferSize = mFrameCount * mFrameSize; |
| // 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. |
| nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; |
| nsecs_t lastWarning = 0; |
| bool longStandbyExit = false; |
| uint32_t activeSleepTime = activeSleepTimeUs(); |
| uint32_t idleSleepTime = idleSleepTimeUs(); |
| uint32_t sleepTime = idleSleepTime; |
| uint32_t sleepTimeShift = 0; |
| Vector< sp<EffectChain> > effectChains; |
| #ifdef DEBUG_CPU_USAGE |
| ThreadCpuUsage cpu; |
| const CentralTendencyStatistics& stats = cpu.statistics(); |
| #endif |
| |
| acquireWakeLock(); |
| |
| while (!exitPending()) |
| { |
| #ifdef DEBUG_CPU_USAGE |
| cpu.sampleAndEnable(); |
| unsigned n = stats.n(); |
| // cpu.elapsed() is expensive, so don't call it every loop |
| if ((n & 127) == 1) { |
| long long elapsed = cpu.elapsed(); |
| if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { |
| double perLoop = elapsed / (double) n; |
| double perLoop100 = perLoop * 0.01; |
| double mean = stats.mean(); |
| double stddev = stats.stddev(); |
| double minimum = stats.minimum(); |
| double maximum = stats.maximum(); |
| cpu.resetStatistics(); |
| LOGI("CPU usage over past %.1f secs (%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", |
| elapsed * .000000001, n, perLoop * .000001, |
| mean * .001, |
| stddev * .001, |
| minimum * .001, |
| maximum * .001, |
| mean / perLoop100, |
| stddev / perLoop100, |
| minimum / perLoop100, |
| maximum / perLoop100); |
| } |
| } |
| #endif |
| processConfigEvents(); |
| |
| mixerStatus = MIXER_IDLE; |
| { // scope for mLock |
| |
| Mutex::Autolock _l(mLock); |
| |
| if (checkForNewParameters_l()) { |
| mixBufferSize = mFrameCount * mFrameSize; |
| // 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(mFrameCount) / mSampleRate * 15; |
| activeSleepTime = activeSleepTimeUs(); |
| idleSleepTime = idleSleepTimeUs(); |
| } |
| |
| const SortedVector< wp<Track> >& activeTracks = mActiveTracks; |
| |
| // put audio hardware into standby after short delay |
| if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || |
| mSuspended)) { |
| if (!mStandby) { |
| ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); |
| mOutput->stream->common.standby(&mOutput->stream->common); |
| mStandby = true; |
| mBytesWritten = 0; |
| } |
| |
| if (!activeTracks.size() && mConfigEvents.isEmpty()) { |
| // we're about to wait, flush the binder command buffer |
| IPCThreadState::self()->flushCommands(); |
| |
| if (exitPending()) break; |
| |
| releaseWakeLock_l(); |
| // wait until we have something to do... |
| ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); |
| mWaitWorkCV.wait(mLock); |
| ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); |
| acquireWakeLock_l(); |
| |
| if (mMasterMute == false) { |
| char value[PROPERTY_VALUE_MAX]; |
| property_get("ro.audio.silent", value, "0"); |
| if (atoi(value)) { |
| LOGD("Silence is golden"); |
| setMasterMute(true); |
| } |
| } |
| |
| standbyTime = systemTime() + kStandbyTimeInNsecs; |
| sleepTime = idleSleepTime; |
| sleepTimeShift = 0; |
| continue; |
| } |
| } |
| |
| mixerStatus = prepareTracks_l(activeTracks, &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(mixerStatus == MIXER_TRACKS_READY)) { |
| // mix buffers... |
| mAudioMixer->process(); |
| sleepTime = 0; |
| // increase sleep time progressively when application underrun condition clears |
| if (sleepTimeShift > 0) { |
| sleepTimeShift--; |
| } |
| standbyTime = systemTime() + kStandbyTimeInNsecs; |
| //TODO: delay standby when effects have a tail |
| } else { |
| // 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 (mixerStatus == 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 || |
| (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { |
| memset (mMixBuffer, 0, mixBufferSize); |
| sleepTime = 0; |
| ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); |
| } |
| // TODO add standby time extension fct of effect tail |
| } |
| |
| if (mSuspended) { |
| sleepTime = suspendSleepTimeUs(); |
| } |
| // sleepTime == 0 means we must write to audio hardware |
| if (sleepTime == 0) { |
| for (size_t i = 0; i < effectChains.size(); i ++) { |
| effectChains[i]->process_l(); |
| } |
| // enable changes in effect chain |
| unlockEffectChains(effectChains); |
| mLastWriteTime = systemTime(); |
| mInWrite = true; |
| mBytesWritten += mixBufferSize; |
| |
| int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); |
| if (bytesWritten < 0) mBytesWritten -= mixBufferSize; |
| mNumWrites++; |
| mInWrite = false; |
| nsecs_t now = systemTime(); |
| nsecs_t delta = now - mLastWriteTime; |
| if (!mStandby && delta > maxPeriod) { |
| mNumDelayedWrites++; |
| if ((now - lastWarning) > kWarningThrottleNs) { |
| LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", |
| ns2ms(delta), mNumDelayedWrites, this); |
| lastWarning = now; |
| } |
| if (mStandby) { |
| longStandbyExit = true; |
| } |
| } |
| mStandby = false; |
| } else { |
| // enable changes in effect chain |
| unlockEffectChains(effectChains); |
| usleep(sleepTime); |
| } |
| |
| // finally let go of all our tracks, without the lock held |
| // since we can't guarantee the destructors won't acquire that |
| // same lock. |
| tracksToRemove.clear(); |
| |
| // Effect chains will be actually deleted here if they were removed from |
| // mEffectChains list during mixing or effects processing |
| effectChains.clear(); |
| } |
| |
| if (!mStandby) { |
| mOutput->stream->common.standby(&mOutput->stream->common); |
| } |
| |
| releaseWakeLock(); |
| |
| ALOGV("MixerThread %p exiting", this); |
| return false; |
| } |
| |
| // prepareTracks_l() must be called with ThreadBase::mLock held |
| uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) |
| { |
| |
| uint32_t mixerStatus = MIXER_IDLE; |
| // find out which tracks need to be processed |
| size_t count = activeTracks.size(); |
| size_t mixedTracks = 0; |
| size_t tracksWithEffect = 0; |
| |
| float masterVolume = mMasterVolume; |
| bool masterMute = mMasterMute; |
| |
| if (masterMute) { |
| masterVolume = 0; |
| } |
| // Delegate master volume control to effect in output mix effect chain if needed |
| sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); |
| if (chain != 0) { |
| uint32_t v = (uint32_t)(masterVolume * (1 << 24)); |
| chain->setVolume_l(&v, &v); |
| masterVolume = (float)((v + (1 << 23)) >> 24); |
| chain.clear(); |
| } |
| |
| for (size_t i=0 ; i<count ; i++) { |
| sp<Track> t = activeTracks[i].promote(); |
| if (t == 0) continue; |
| |
| 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 |
| mAudioMixer->setActiveTrack(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 (track->mRetryCount >= kMaxTrackRetries) meaning the track was mixed |
| // during last round |
| uint32_t minFrames = 1; |
| if (!track->isStopped() && !track->isPausing() && |
| (track->mRetryCount >= kMaxTrackRetries)) { |
| if (t->sampleRate() == (int)mSampleRate) { |
| minFrames = mFrameCount; |
| } else { |
| minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1; |
| } |
| } |
| if ((cblk->framesReady() >= minFrames) && track->isReady() && |
| !track->isPaused() && !track->isTerminated()) |
| { |
| //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->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 { |
| LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", |
| track->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(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->type()].mute) { |
| vl = vr = va = 0; |
| if (track->isPausing()) { |
| track->setPaused(); |
| } |
| } else { |
| |
| // read original volumes with volume control |
| float typeVolume = mStreamTypes[track->type()].volume; |
| float v = masterVolume * typeVolume; |
| vl = (uint32_t)(v * cblk->volume[0]) << 12; |
| vr = (uint32_t)(v * cblk->volume[1]) << 12; |
| |
| va = (uint32_t)(v * cblk->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 |
| int16_t left, right, aux; |
| uint32_t v_clamped = (vl + (1 << 11)) >> 12; |
| if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; |
| left = int16_t(v_clamped); |
| v_clamped = (vr + (1 << 11)) >> 12; |
| if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; |
| right = int16_t(v_clamped); |
| |
| if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; |
| aux = int16_t(va); |
| |
| // XXX: these things DON'T need to be done each time |
| mAudioMixer->setBufferProvider(track); |
| mAudioMixer->enable(); |
| |
| mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); |
| mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); |
| mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); |
| mAudioMixer->setParameter( |
| AudioMixer::TRACK, |
| AudioMixer::FORMAT, (void *)track->format()); |
| mAudioMixer->setParameter( |
| AudioMixer::TRACK, |
| AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); |
| mAudioMixer->setParameter( |
| AudioMixer::RESAMPLE, |
| AudioMixer::SAMPLE_RATE, |
| (void *)(cblk->sampleRate)); |
| mAudioMixer->setParameter( |
| AudioMixer::TRACK, |
| AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); |
| mAudioMixer->setParameter( |
| AudioMixer::TRACK, |
| AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); |
| |
| // reset retry count |
| track->mRetryCount = kMaxTrackRetries; |
| mixerStatus = MIXER_TRACKS_READY; |
| } else { |
| //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); |
| if (track->isStopped()) { |
| track->reset(); |
| } |
| if (track->isTerminated() || track->isStopped() || track->isPaused()) { |
| // We have consumed all the buffers of this track. |
| // Remove it from the list of active tracks. |
| tracksToRemove->add(track); |
| } else { |
| // No buffers for this track. Give it a few chances to |
| // fill a buffer, then remove it from active list. |
| if (--(track->mRetryCount) <= 0) { |
| ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); |
| tracksToRemove->add(track); |
| // indicate to client process that the track was disabled because of underrun |
| android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); |
| } else if (mixerStatus != MIXER_TRACKS_READY) { |
| mixerStatus = MIXER_TRACKS_ENABLED; |
| } |
| } |
| mAudioMixer->disable(); |
| } |
| } |
| |
| // remove all the tracks that need to be... |
| count = tracksToRemove->size(); |
| if (CC_UNLIKELY(count)) { |
| for (size_t i=0 ; i<count ; i++) { |
| const sp<Track>& 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) { |
| memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); |
| } |
| |
| return mixerStatus; |
| } |
| |
| void AudioFlinger::MixerThread::invalidateTracks(int 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<Track> t = mTracks[i]; |
| if (t->type() == streamType) { |
| android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); |
| t->mCblk->cv.signal(); |
| } |
| } |
| } |
| |
| void AudioFlinger::PlaybackThread::setStreamValid(int streamType, bool valid) |
| { |
| ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", |
| this, streamType, valid); |
| Mutex::Autolock _l(mLock); |
| |
| mStreamTypes[streamType].valid = valid; |
| } |
| |
| // getTrackName_l() must be called with ThreadBase::mLock held |
| int AudioFlinger::MixerThread::getTrackName_l() |
| { |
| return mAudioMixer->getTrackName(); |
| } |
| |
| // 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() |
| { |
| 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::keySamplingRate), value) == NO_ERROR) { |
| reconfig = true; |
| } |
| if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { |
| if (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) { |
| // when changing the audio output device, call addBatteryData to notify |
| // the change |
| if ((int)mDevice != value) { |
| uint32_t params = 0; |
| // check whether speaker is on |
| if (value & AUDIO_DEVICE_OUT_SPEAKER) { |
| params |= IMediaPlayerService::kBatteryDataSpeakerOn; |
| } |
| |
| int 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); |
| } |
| } |
| |
| // forward device change to effects that have requested to be |
| // aware of attached audio device. |
| mDevice = (uint32_t)value; |
| for (size_t i = 0; i < mEffectChains.size(); i++) { |
| mEffectChains[i]->setDevice_l(mDevice); |
| } |
| } |
| |
| 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; |
| readOutputParameters(); |
| mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); |
| for (size_t i = 0; i < mTracks.size() ; i++) { |
| int name = getTrackName_l(); |
| 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(); |
| } |
| } |
| sendConfigEvent_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; |
| } |
| |
| status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& 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()); |
| return NO_ERROR; |
| } |
| |
| uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() |
| { |
| return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; |
| } |
| |
| uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() |
| { |
| return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) |
| : PlaybackThread(audioFlinger, output, id, device) |
| { |
| mType = ThreadBase::DIRECT; |
| } |
| |
| AudioFlinger::DirectOutputThread::~DirectOutputThread() |
| { |
| } |
| |
| static inline |
| int32_t mul(int16_t in, int16_t v) |
| { |
| #if defined(__arm__) && !defined(__thumb__) |
| int32_t out; |
| asm( "smulbb %[out], %[in], %[v] \n" |
| : [out]"=r"(out) |
| : [in]"%r"(in), [v]"r"(v) |
| : ); |
| return out; |
| #else |
| return in * int32_t(v); |
| #endif |
| } |
| |
| void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) |
| { |
| // Do not apply volume on compressed audio |
| if (!audio_is_linear_pcm(mFormat)) { |
| return; |
| } |
| |
| // convert to signed 16 bit before volume calculation |
| if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { |
| size_t count = mFrameCount * mChannelCount; |
| uint8_t *src = (uint8_t *)mMixBuffer + count-1; |
| int16_t *dst = mMixBuffer + count-1; |
| while(count--) { |
| *dst-- = (int16_t)(*src--^0x80) << 8; |
| } |
| } |
| |
| size_t frameCount = mFrameCount; |
| int16_t *out = mMixBuffer; |
| if (ramp) { |
| if (mChannelCount == 1) { |
| int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; |
| int32_t vlInc = d / (int32_t)frameCount; |
| int32_t vl = ((int32_t)mLeftVolShort << 16); |
| do { |
| out[0] = clamp16(mul(out[0], vl >> 16) >> 12); |
| out++; |
| vl += vlInc; |
| } while (--frameCount); |
| |
| } else { |
| int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; |
| int32_t vlInc = d / (int32_t)frameCount; |
| d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; |
| int32_t vrInc = d / (int32_t)frameCount; |
| int32_t vl = ((int32_t)mLeftVolShort << 16); |
| int32_t vr = ((int32_t)mRightVolShort << 16); |
| do { |
| out[0] = clamp16(mul(out[0], vl >> 16) >> 12); |
| out[1] = clamp16(mul(out[1], vr >> 16) >> 12); |
| out += 2; |
| vl += vlInc; |
| vr += vrInc; |
| } while (--frameCount); |
| } |
| } else { |
| if (mChannelCount == 1) { |
| do { |
| out[0] = clamp16(mul(out[0], leftVol) >> 12); |
| out++; |
| } while (--frameCount); |
| } else { |
| do { |
| out[0] = clamp16(mul(out[0], leftVol) >> 12); |
| out[1] = clamp16(mul(out[1], rightVol) >> 12); |
| out += 2; |
| } while (--frameCount); |
| } |
| } |
| |
| // convert back to unsigned 8 bit after volume calculation |
| if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { |
| size_t count = mFrameCount * mChannelCount; |
| int16_t *src = mMixBuffer; |
| uint8_t *dst = (uint8_t *)mMixBuffer; |
| while(count--) { |
| *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; |
| } |
| } |
| |
| mLeftVolShort = leftVol; |
| mRightVolShort = rightVol; |
| } |
| |
| bool AudioFlinger::DirectOutputThread::threadLoop() |
| { |
| uint32_t mixerStatus = MIXER_IDLE; |
| sp<Track> trackToRemove; |
| sp<Track> activeTrack; |
| nsecs_t standbyTime = systemTime(); |
| int8_t *curBuf; |
| size_t mixBufferSize = mFrameCount*mFrameSize; |
| uint32_t activeSleepTime = activeSleepTimeUs(); |
| uint32_t idleSleepTime = idleSleepTimeUs(); |
| uint32_t sleepTime = idleSleepTime; |
| // use shorter standby delay as on normal output to release |
| // hardware resources as soon as possible |
| nsecs_t standbyDelay = microseconds(activeSleepTime*2); |
| |
| acquireWakeLock(); |
| |
| while (!exitPending()) |
| { |
| bool rampVolume; |
| uint16_t leftVol; |
| uint16_t rightVol; |
| Vector< sp<EffectChain> > effectChains; |
| |
| processConfigEvents(); |
| |
| mixerStatus = MIXER_IDLE; |
| |
| { // scope for the mLock |
| |
| Mutex::Autolock _l(mLock); |
| |
| if (checkForNewParameters_l()) { |
| mixBufferSize = mFrameCount*mFrameSize; |
| activeSleepTime = activeSleepTimeUs(); |
| idleSleepTime = idleSleepTimeUs(); |
| standbyDelay = microseconds(activeSleepTime*2); |
| } |
| |
| // put audio hardware into standby after short delay |
| if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || |
| mSuspended)) { |
| // wait until we have something to do... |
| if (!mStandby) { |
| ALOGV("Audio hardware entering standby, mixer %p\n", this); |
| mOutput->stream->common.standby(&mOutput->stream->common); |
| mStandby = true; |
| mBytesWritten = 0; |
| } |
| |
| if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { |
| // we're about to wait, flush the binder command buffer |
| IPCThreadState::self()->flushCommands(); |
| |
| if (exitPending()) break; |
| |
| releaseWakeLock_l(); |
| ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); |
| mWaitWorkCV.wait(mLock); |
| ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); |
| acquireWakeLock_l(); |
| |
| if (mMasterMute == false) { |
| char value[PROPERTY_VALUE_MAX]; |
| property_get("ro.audio.silent", value, "0"); |
| if (atoi(value)) { |
| LOGD("Silence is golden"); |
| setMasterMute(true); |
| } |
| } |
| |
| standbyTime = systemTime() + standbyDelay; |
| sleepTime = idleSleepTime; |
| continue; |
| } |
| } |
| |
| effectChains = mEffectChains; |
| |
| // find out which tracks need to be processed |
| if (mActiveTracks.size() != 0) { |
| sp<Track> t = mActiveTracks[0].promote(); |
| if (t == 0) continue; |
| |
| 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 |
| if (cblk->framesReady() && 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; |
| mLeftVolShort = mRightVolShort = 0; |
| if (track->mState == TrackBase::RESUMING) { |
| track->mState = TrackBase::ACTIVE; |
| rampVolume = true; |
| } |
| } else if (cblk->server != 0) { |
| // If the track is stopped before the first frame was mixed, |
| // do not apply ramp |
| rampVolume = true; |
| } |
| // compute volume for this track |
| float left, right; |
| if (track->isMuted() || mMasterMute || track->isPausing() || |
| mStreamTypes[track->type()].mute) { |
| left = right = 0; |
| if (track->isPausing()) { |
| track->setPaused(); |
| } |
| } else { |
| float typeVolume = mStreamTypes[track->type()].volume; |
| float v = mMasterVolume * typeVolume; |
| float v_clamped = v * cblk->volume[0]; |
| if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; |
| left = v_clamped/MAX_GAIN; |
| v_clamped = v * cblk->volume[1]; |
| if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; |
| right = v_clamped/MAX_GAIN; |
| } |
| |
| if (left != mLeftVolFloat || right != mRightVolFloat) { |
| mLeftVolFloat = left; |
| mRightVolFloat = right; |
| |
| // If audio HAL implements volume control, |
| // force software volume to nominal value |
| if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { |
| left = 1.0f; |
| right = 1.0f; |
| } |
| |
| // 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 (!effectChains.isEmpty()) { |
| // Do not ramp volume if volume is controlled by effect |
| if(effectChains[0]->setVolume_l(&vl, &vr)) { |
| rampVolume = false; |
| } |
| } |
| |
| // Convert volumes from 8.24 to 4.12 format |
| uint32_t v_clamped = (vl + (1 << 11)) >> 12; |
| if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; |
| leftVol = (uint16_t)v_clamped; |
| v_clamped = (vr + (1 << 11)) >> 12; |
| if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; |
| rightVol = (uint16_t)v_clamped; |
| } else { |
| leftVol = mLeftVolShort; |
| rightVol = mRightVolShort; |
| rampVolume = false; |
| } |
| |
| // reset retry count |
| track->mRetryCount = kMaxTrackRetriesDirect; |
| activeTrack = t; |
| mixerStatus = MIXER_TRACKS_READY; |
| } else { |
| //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); |
| if (track->isStopped()) { |
| track->reset(); |
| } |
| if (track->isTerminated() || track->isStopped() || track->isPaused()) { |
| // We have consumed all the buffers of this track. |
| // Remove it from the list of active tracks. |
| 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; |
| } |
| } |
| } |
| } |
| |
| // remove all the tracks that need to be... |
| if (CC_UNLIKELY(trackToRemove != 0)) { |
| mActiveTracks.remove(trackToRemove); |
| if (!effectChains.isEmpty()) { |
| ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), |
| trackToRemove->sessionId()); |
| effectChains[0]->decActiveTrackCnt(); |
| } |
| if (trackToRemove->isTerminated()) { |
| removeTrack_l(trackToRemove); |
| } |
| } |
| |
| lockEffectChains_l(effectChains); |
| } |
| |
| if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { |
| AudioBufferProvider::Buffer buffer; |
| size_t frameCount = mFrameCount; |
| curBuf = (int8_t *)mMixBuffer; |
| // output audio to hardware |
| while (frameCount) { |
| buffer.frameCount = frameCount; |
| activeTrack->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; |
| activeTrack->releaseBuffer(&buffer); |
| } |
| sleepTime = 0; |
| standbyTime = systemTime() + standbyDelay; |
| } else { |
| if (sleepTime == 0) { |
| if (mixerStatus == MIXER_TRACKS_ENABLED) { |
| sleepTime = activeSleepTime; |
| } else { |
| sleepTime = idleSleepTime; |
| } |
| } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { |
| memset (mMixBuffer, 0, mFrameCount * mFrameSize); |
| sleepTime = 0; |
| } |
| } |
| |
| if (mSuspended) { |
| sleepTime = suspendSleepTimeUs(); |
| } |
| // sleepTime == 0 means we must write to audio hardware |
| if (sleepTime == 0) { |
| if (mixerStatus == MIXER_TRACKS_READY) { |
| applyVolume(leftVol, rightVol, rampVolume); |
| } |
| for (size_t i = 0; i < effectChains.size(); i ++) { |
| effectChains[i]->process_l(); |
| } |
| unlockEffectChains(effectChains); |
| |
| mLastWriteTime = systemTime(); |
| mInWrite = true; |
| mBytesWritten += mixBufferSize; |
| int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); |
| if (bytesWritten < 0) mBytesWritten -= mixBufferSize; |
| mNumWrites++; |
| mInWrite = false; |
| mStandby = false; |
| } else { |
| unlockEffectChains(effectChains); |
| usleep(sleepTime); |
| } |
| |
| // finally let go of removed track, without the lock held |
| // since we can't guarantee the destructors won't acquire that |
| // same lock. |
| trackToRemove.clear(); |
| activeTrack.clear(); |
| |
| // Effect chains will be actually deleted here if they were removed from |
| // mEffectChains list during mixing or effects processing |
| effectChains.clear(); |
| } |
| |
| if (!mStandby) { |
| mOutput->stream->common.standby(&mOutput->stream->common); |
| } |
| |
| releaseWakeLock(); |
| |
| ALOGV("DirectOutputThread %p exiting", this); |
| return false; |
| } |
| |
| // getTrackName_l() must be called with ThreadBase::mLock held |
| int AudioFlinger::DirectOutputThread::getTrackName_l() |
| { |
| 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(); |
| sendConfigEvent_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() |
| { |
| uint32_t time; |
| if (audio_is_linear_pcm(mFormat)) { |
| time = PlaybackThread::activeSleepTimeUs(); |
| } else { |
| time = 10000; |
| } |
| return time; |
| } |
| |
| uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() |
| { |
| 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() |
| { |
| uint32_t time; |
| if (audio_is_linear_pcm(mFormat)) { |
| time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); |
| } else { |
| time = 10000; |
| } |
| return time; |
| } |
| |
| |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) |
| : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) |
| { |
| mType = ThreadBase::DUPLICATING; |
| addOutputTrack(mainThread); |
| } |
| |
| AudioFlinger::DuplicatingThread::~DuplicatingThread() |
| { |
| for (size_t i = 0; i < mOutputTracks.size(); i++) { |
| mOutputTracks[i]->destroy(); |
| } |
| mOutputTracks.clear(); |
| } |
| |
| bool AudioFlinger::DuplicatingThread::threadLoop() |
| { |
| Vector< sp<Track> > tracksToRemove; |
| uint32_t mixerStatus = MIXER_IDLE; |
| nsecs_t standbyTime = systemTime(); |
| size_t mixBufferSize = mFrameCount*mFrameSize; |
| SortedVector< sp<OutputTrack> > outputTracks; |
| uint32_t writeFrames = 0; |
| uint32_t activeSleepTime = activeSleepTimeUs(); |
| uint32_t idleSleepTime = idleSleepTimeUs(); |
| uint32_t sleepTime = idleSleepTime; |
| Vector< sp<EffectChain> > effectChains; |
| |
| acquireWakeLock(); |
| |
| while (!exitPending()) |
| { |
| processConfigEvents(); |
| |
| mixerStatus = MIXER_IDLE; |
| { // scope for the mLock |
| |
| Mutex::Autolock _l(mLock); |
| |
| if (checkForNewParameters_l()) { |
| mixBufferSize = mFrameCount*mFrameSize; |
| updateWaitTime(); |
| activeSleepTime = activeSleepTimeUs(); |
| idleSleepTime = idleSleepTimeUs(); |
| } |
| |
| const SortedVector< wp<Track> >& activeTracks = mActiveTracks; |
| |
| for (size_t i = 0; i < mOutputTracks.size(); i++) { |
| outputTracks.add(mOutputTracks[i]); |
| } |
| |
| // put audio hardware into standby after short delay |
| if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || |
| mSuspended)) { |
| if (!mStandby) { |
| for (size_t i = 0; i < outputTracks.size(); i++) { |
| outputTracks[i]->stop(); |
| } |
| mStandby = true; |
| mBytesWritten = 0; |
| } |
| |
| if (!activeTracks.size() && mConfigEvents.isEmpty()) { |
| // we're about to wait, flush the binder command buffer |
| IPCThreadState::self()->flushCommands(); |
| outputTracks.clear(); |
| |
| if (exitPending()) break; |
| |
| releaseWakeLock_l(); |
| ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); |
| mWaitWorkCV.wait(mLock); |
| ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); |
| acquireWakeLock_l(); |
| |
| if (mMasterMute == false) { |
| char value[PROPERTY_VALUE_MAX]; |
| property_get("ro.audio.silent", value, "0"); |
| if (atoi(value)) { |
| LOGD("Silence is golden"); |
| setMasterMute(true); |
| } |
| } |
| |
| standbyTime = systemTime() + kStandbyTimeInNsecs; |
| sleepTime = idleSleepTime; |
| continue; |
| } |
| } |
| |
| mixerStatus = prepareTracks_l(activeTracks, &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(mixerStatus == MIXER_TRACKS_READY)) { |
| // mix buffers... |
| if (outputsReady(outputTracks)) { |
| mAudioMixer->process(); |
| } else { |
| memset(mMixBuffer, 0, mixBufferSize); |
| } |
| sleepTime = 0; |
| writeFrames = mFrameCount; |
| } else { |
| if (sleepTime == 0) { |
| if (mixerStatus == MIXER_TRACKS_ENABLED) { |
| sleepTime = activeSleepTime; |
| } else { |
| sleepTime = idleSleepTime; |
| } |
| } else if (mBytesWritten != 0) { |
| // flush remaining overflow buffers in output tracks |
| for (size_t i = 0; i < outputTracks.size(); i++) { |
| if (outputTracks[i]->isActive()) { |
| sleepTime = 0; |
| writeFrames = 0; |
| memset(mMixBuffer, 0, mixBufferSize); |
| break; |
| } |
| } |
| } |
| } |
| |
| if (mSuspended) { |
| sleepTime = suspendSleepTimeUs(); |
| } |
| // sleepTime == 0 means we must write to audio hardware |
| if (sleepTime == 0) { |
| for (size_t i = 0; i < effectChains.size(); i ++) { |
| effectChains[i]->process_l(); |
| } |
| // enable changes in effect chain |
| unlockEffectChains(effectChains); |
| |
| standbyTime = systemTime() + kStandbyTimeInNsecs; |
| for (size_t i = 0; i < outputTracks.size(); i++) { |
| outputTracks[i]->write(mMixBuffer, writeFrames); |
| } |
| mStandby = false; |
| mBytesWritten += mixBufferSize; |
| } else { |
| // enable changes in effect chain |
| unlockEffectChains(effectChains); |
| usleep(sleepTime); |
| } |
| |
| // finally let go of all our tracks, without the lock held |
| // since we can't guarantee the destructors won't acquire that |
| // same lock. |
| tracksToRemove.clear(); |
| outputTracks.clear(); |
| |
| // Effect chains will be actually deleted here if they were removed from |
| // mEffectChains list during mixing or effects processing |
| effectChains.clear(); |
| } |
| |
| releaseWakeLock(); |
| |
| return false; |
| } |
| |
| void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) |
| { |
| int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); |
| OutputTrack *outputTrack = new OutputTrack((ThreadBase *)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(); |
| } |
| } |
| |
| void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) |
| { |
| Mutex::Autolock _l(mLock); |
| for (size_t i = 0; i < mOutputTracks.size(); i++) { |
| if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { |
| mOutputTracks[i]->destroy(); |
| mOutputTracks.removeAt(i); |
| updateWaitTime(); |
| return; |
| } |
| } |
| ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); |
| } |
| |
| void AudioFlinger::DuplicatingThread::updateWaitTime() |
| { |
| mWaitTimeMs = UINT_MAX; |
| for (size_t i = 0; i < mOutputTracks.size(); i++) { |
| sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); |
| if (strong != NULL) { |
| uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); |
| if (waitTimeMs < mWaitTimeMs) { |
| mWaitTimeMs = waitTimeMs; |
| } |
| } |
| } |
| } |
| |
| |
| bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) |
| { |
| for (size_t i = 0; i < outputTracks.size(); i++) { |
| sp <ThreadBase> thread = outputTracks[i]->thread().promote(); |
| if (thread == 0) { |
| LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); |
| return false; |
| } |
| PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); |
| 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() |
| { |
| return (mWaitTimeMs * 1000) / 2; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| // TrackBase constructor must be called with AudioFlinger::mLock held |
| AudioFlinger::ThreadBase::TrackBase::TrackBase( |
| const wp<ThreadBase>& thread, |
| const sp<Client>& client, |
| uint32_t sampleRate, |
| uint32_t format, |
| uint32_t channelMask, |
| int frameCount, |
| uint32_t flags, |
| const sp<IMemory>& sharedBuffer, |
| int sessionId) |
| : RefBase(), |
| mThread(thread), |
| mClient(client), |
| mCblk(0), |
| mFrameCount(0), |
| mState(IDLE), |
| mClientTid(-1), |
| mFormat(format), |
| mFlags(flags & ~SYSTEM_FLAGS_MASK), |
| mSessionId(sessionId) |
| { |
| ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); |
| |
| // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); |
| size_t size = sizeof(audio_track_cblk_t); |
| uint8_t channelCount = popcount(channelMask); |
| size_t bufferSize = frameCount*channelCount*sizeof(int16_t); |
| if (sharedBuffer == 0) { |
| size += bufferSize; |
| } |
| |
| if (client != NULL) { |
| mCblkMemory = client->heap()->allocate(size); |
| if (mCblkMemory != 0) { |
| mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); |
| if (mCblk) { // construct the shared structure in-place. |
| new(mCblk) audio_track_cblk_t(); |
| // clear all buffers |
| mCblk->frameCount = frameCount; |
| mCblk->sampleRate = sampleRate; |
| mChannelCount = channelCount; |
| mChannelMask = channelMask; |
| if (sharedBuffer == 0) { |
| 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; |
| } else { |
| mBuffer = sharedBuffer->pointer(); |
| } |
| mBufferEnd = (uint8_t *)mBuffer + bufferSize; |
| } |
| } else { |
| LOGE("not enough memory for AudioTrack size=%u", size); |
| client->heap()->dump("AudioTrack"); |
| return; |
| } |
| } else { |
| mCblk = (audio_track_cblk_t *)(new uint8_t[size]); |
| if (mCblk) { // construct the shared structure in-place. |
| new(mCblk) audio_track_cblk_t(); |
| // clear all buffers |
| mCblk->frameCount = frameCount; |
| mCblk->sampleRate = sampleRate; |
| 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) { |
| mCblk->~audio_track_cblk_t(); // destroy our shared-structure. |
| if (mClient == NULL) { |
| delete mCblk; |
| } |
| } |
| mCblkMemory.clear(); // and free the shared memory |
| if (mClient != NULL) { |
| Mutex::Autolock _l(mClient->audioFlinger()->mLock); |
| mClient.clear(); |
| } |
| } |
| |
| void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) |
| { |
| buffer->raw = NULL; |
| mFrameCount = buffer->frameCount; |
| 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"); |
| mFlags |= STEPSERVER_FAILED; |
| } |
| 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; |
| mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); |
| ALOGV("TrackBase::reset"); |
| } |
| |
| sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const |
| { |
| return mCblkMemory; |
| } |
| |
| int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { |
| return (int)mCblk->sampleRate; |
| } |
| |
| int AudioFlinger::ThreadBase::TrackBase::channelCount() const { |
| return (const int)mChannelCount; |
| } |
| |
| uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { |
| return mChannelMask; |
| } |
| |
| void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { |
| audio_track_cblk_t* cblk = this->cblk(); |
| int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; |
| int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; |
| |
| // Check validity of returned pointer in case the track control block would have been corrupted. |
| if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || |
| ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { |
| LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ |
| server %d, serverBase %d, user %d, userBase %d", |
| bufferStart, bufferEnd, mBuffer, mBufferEnd, |
| cblk->server, cblk->serverBase, cblk->user, cblk->userBase); |
| return 0; |
| } |
| |
| return bufferStart; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| // Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held |
| AudioFlinger::PlaybackThread::Track::Track( |
| const wp<ThreadBase>& thread, |
| const sp<Client>& client, |
| int streamType, |
| uint32_t sampleRate, |
| uint32_t format, |
| uint32_t channelMask, |
| int frameCount, |
| const sp<IMemory>& sharedBuffer, |
| int sessionId) |
| : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), |
| mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), |
| mAuxEffectId(0), mHasVolumeController(false) |
| { |
| if (mCblk != NULL) { |
| sp<ThreadBase> baseThread = thread.promote(); |
| if (baseThread != 0) { |
| PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); |
| mName = playbackThread->getTrackName_l(); |
| mMainBuffer = playbackThread->mixBuffer(); |
| } |
| ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); |
| if (mName < 0) { |
| LOGE("no more track names available"); |
| } |
| mVolume[0] = 1.0f; |
| mVolume[1] = 1.0f; |
| mStreamType = streamType; |
| // 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 |
| mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); |
| } |
| } |
| |
| AudioFlinger::PlaybackThread::Track::~Track() |
| { |
| ALOGV("PlaybackThread::Track destructor"); |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| Mutex::Autolock _l(thread->mLock); |
| mState = TERMINATED; |
| } |
| } |
| |
| 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 |
| // desctructor 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<Track> keep(this); |
| { // scope for mLock |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| if (!isOutputTrack()) { |
| if (mState == ACTIVE || mState == RESUMING) { |
| AudioSystem::stopOutput(thread->id(), |
| (audio_stream_type_t)mStreamType, |
| mSessionId); |
| |
| // to track the speaker usage |
| addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); |
| } |
| AudioSystem::releaseOutput(thread->id()); |
| } |
| Mutex::Autolock _l(thread->mLock); |
| PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); |
| playbackThread->destroyTrack_l(this); |
| } |
| } |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) |
| { |
| snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n", |
| mName - AudioMixer::TRACK0, |
| (mClient == NULL) ? getpid() : mClient->pid(), |
| mStreamType, |
| mFormat, |
| mChannelMask, |
| mSessionId, |
| mFrameCount, |
| mState, |
| mMute, |
| mFillingUpStatus, |
| mCblk->sampleRate, |
| mCblk->volume[0], |
| mCblk->volume[1], |
| mCblk->server, |
| mCblk->user, |
| (int)mMainBuffer, |
| (int)mAuxBuffer); |
| } |
| |
| status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) |
| { |
| 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 (mFlags & TrackBase::STEPSERVER_FAILED) { |
| if (!step()) goto getNextBuffer_exit; |
| ALOGV("stepServer recovered"); |
| mFlags &= ~TrackBase::STEPSERVER_FAILED; |
| } |
| |
| 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 (s + framesReq > bufferEnd) { |
| framesReq = bufferEnd - s; |
| } |
| |
| buffer->raw = getBuffer(s, framesReq); |
| if (buffer->raw == NULL) goto getNextBuffer_exit; |
| |
| 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; |
| } |
| |
| bool AudioFlinger::PlaybackThread::Track::isReady() const { |
| if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; |
| |
| if (mCblk->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() |
| { |
| status_t status = NO_ERROR; |
| ALOGV("start(%d), calling thread %d session %d", |
| mName, IPCThreadState::self()->getCallingPid(), mSessionId); |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| Mutex::Autolock _l(thread->mLock); |
| int 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(), |
| (audio_stream_type_t)mStreamType, |
| mSessionId); |
| thread->mLock.lock(); |
| |
| // to track the speaker usage |
| if (status == NO_ERROR) { |
| addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); |
| } |
| } |
| if (status == NO_ERROR) { |
| PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); |
| playbackThread->addTrack_l(this); |
| } else { |
| mState = state; |
| } |
| } else { |
| status = BAD_VALUE; |
| } |
| return status; |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::stop() |
| { |
| ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| Mutex::Autolock _l(thread->mLock); |
| int state = mState; |
| if (mState > STOPPED) { |
| mState = STOPPED; |
| // If the track is not active (PAUSED and buffers full), flush buffers |
| PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); |
| if (playbackThread->mActiveTracks.indexOf(this) < 0) { |
| reset(); |
| } |
| ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); |
| } |
| if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { |
| thread->mLock.unlock(); |
| AudioSystem::stopOutput(thread->id(), |
| (audio_stream_type_t)mStreamType, |
| mSessionId); |
| thread->mLock.lock(); |
| |
| // to track the speaker usage |
| addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); |
| } |
| } |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::pause() |
| { |
| ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); |
| sp<ThreadBase> 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(), |
| (audio_stream_type_t)mStreamType, |
| mSessionId); |
| thread->mLock.lock(); |
| |
| // to track the speaker usage |
| addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); |
| } |
| } |
| } |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::flush() |
| { |
| ALOGV("flush(%d)", mName); |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| Mutex::Autolock _l(thread->mLock); |
| if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { |
| return; |
| } |
| // No point remaining in PAUSED state after a flush => go to |
| // STOPPED state |
| mState = STOPPED; |
| |
| // 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. |
| 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; |
| } |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::mute(bool muted) |
| { |
| mMute = muted; |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) |
| { |
| mVolume[0] = left; |
| mVolume[1] = right; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) |
| { |
| status_t status = DEAD_OBJECT; |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); |
| status = playbackThread->attachAuxEffect(this, EffectId); |
| } |
| return status; |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) |
| { |
| mAuxEffectId = EffectId; |
| mAuxBuffer = buffer; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| // RecordTrack constructor must be called with AudioFlinger::mLock held |
| AudioFlinger::RecordThread::RecordTrack::RecordTrack( |
| const wp<ThreadBase>& thread, |
| const sp<Client>& client, |
| uint32_t sampleRate, |
| uint32_t format, |
| uint32_t channelMask, |
| int frameCount, |
| uint32_t flags, |
| int sessionId) |
| : TrackBase(thread, client, sampleRate, format, |
| channelMask, frameCount, flags, 0, sessionId), |
| mOverflow(false) |
| { |
| if (mCblk != NULL) { |
| 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); |
| } |
| } |
| } |
| |
| AudioFlinger::RecordThread::RecordTrack::~RecordTrack() |
| { |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| AudioSystem::releaseInput(thread->id()); |
| } |
| } |
| |
| status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) |
| { |
| 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 (mFlags & TrackBase::STEPSERVER_FAILED) { |
| if (!step()) goto getNextBuffer_exit; |
| ALOGV("stepServer recovered"); |
| mFlags &= ~TrackBase::STEPSERVER_FAILED; |
| } |
| |
| 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 (s + framesReq > bufferEnd) { |
| framesReq = bufferEnd - s; |
| } |
| |
| buffer->raw = getBuffer(s, framesReq); |
| if (buffer->raw == NULL) goto getNextBuffer_exit; |
| |
| buffer->frameCount = framesReq; |
| return NO_ERROR; |
| } |
| |
| getNextBuffer_exit: |
| buffer->raw = NULL; |
| buffer->frameCount = 0; |
| return NOT_ENOUGH_DATA; |
| } |
| |
| status_t AudioFlinger::RecordThread::RecordTrack::start() |
| { |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| RecordThread *recordThread = (RecordThread *)thread.get(); |
| return recordThread->start(this); |
| } else { |
| return BAD_VALUE; |
| } |
| } |
| |
| void AudioFlinger::RecordThread::RecordTrack::stop() |
| { |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| RecordThread *recordThread = (RecordThread *)thread.get(); |
| recordThread->stop(this); |
| TrackBase::reset(); |
| // Force overerrun condition to avoid false overrun callback until first data is |
| // read from buffer |
| android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); |
| } |
| } |
| |
| void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) |
| { |
| snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", |
| (mClient == NULL) ? getpid() : mClient->pid(), |
| mFormat, |
| mChannelMask, |
| mSessionId, |
| mFrameCount, |
| mState, |
| mCblk->sampleRate, |
| mCblk->server, |
| mCblk->user); |
| } |
| |
| |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( |
| const wp<ThreadBase>& thread, |
| DuplicatingThread *sourceThread, |
| uint32_t sampleRate, |
| uint32_t format, |
| uint32_t channelMask, |
| int frameCount) |
| : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), |
| mActive(false), mSourceThread(sourceThread) |
| { |
| |
| PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); |
| if (mCblk != NULL) { |
| mCblk->flags |= CBLK_DIRECTION_OUT; |
| mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); |
| mCblk->volume[0] = mCblk->volume[1] = 0x1000; |
| 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 { |
| LOGW("Error creating output track on thread %p", playbackThread); |
| } |
| } |
| |
| AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() |
| { |
| clearBufferQueue(); |
| } |
| |
| status_t AudioFlinger::PlaybackThread::OutputTrack::start() |
| { |
| status_t status = Track::start(); |
| 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<ThreadBase> 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 { |
| LOGW ("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)AudioTrack::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<ThreadBase> 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 { |
| LOGW("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 AudioTrack::NO_MORE_BUFFERS; |
| } |
| result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); |
| if (result != NO_ERROR) { |
| return AudioTrack::NO_MORE_BUFFERS; |
| } |
| // read the server count again |
| start_loop_here: |
| framesAvail = cblk->framesAvailable_l(); |
| } |
| } |
| |
| // if (framesAvail < framesReq) { |
| // return AudioTrack::NO_MORE_BUFFERS; |
| // } |
| |
| if (framesReq > framesAvail) { |
| framesReq = framesAvail; |
| } |
| |
| uint32_t u = cblk->user; |
| uint32_t bufferEnd = cblk->userBase + cblk->frameCount; |
| |
| if (u + framesReq > bufferEnd) { |
| 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(); |
| Buffer *pBuffer; |
| |
| for (size_t i = 0; i < size; i++) { |
| pBuffer = mBufferQueue.itemAt(i); |
| delete [] pBuffer->mBuffer; |
| delete pBuffer; |
| } |
| mBufferQueue.clear(); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) |
| : RefBase(), |
| mAudioFlinger(audioFlinger), |
| mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), |
| mPid(pid) |
| { |
| // 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); |
| } |
| |
| const sp<MemoryDealer>& AudioFlinger::Client::heap() const |
| { |
| return mMemoryDealer; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, |
| const sp<IAudioFlingerClient>& client, |
| pid_t pid) |
| : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) |
| { |
| } |
| |
| AudioFlinger::NotificationClient::~NotificationClient() |
| { |
| mClient.clear(); |
| } |
| |
| void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) |
| { |
| sp<NotificationClient> keep(this); |
| { |
| mAudioFlinger->removeNotificationClient(mPid); |
| } |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& 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(); |
| } |
| |
| 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(); |
| } |
| |
| void AudioFlinger::TrackHandle::setVolume(float left, float right) { |
| mTrack->setVolume(left, right); |
| } |
| |
| sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { |
| return mTrack->getCblk(); |
| } |
| |
| status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) |
| { |
| return mTrack->attachAuxEffect(EffectId); |
| } |
| |
| status_t AudioFlinger::TrackHandle::onTransact( |
| uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) |
| { |
| return BnAudioTrack::onTransact(code, data, reply, flags); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| sp<IAudioRecord> AudioFlinger::openRecord( |
| pid_t pid, |
| int input, |
| uint32_t sampleRate, |
| uint32_t format, |
| uint32_t channelMask, |
| int frameCount, |
| uint32_t flags, |
| int *sessionId, |
| status_t *status) |
| { |
| sp<RecordThread::RecordTrack> recordTrack; |
| sp<RecordHandle> recordHandle; |
| sp<Client> client; |
| wp<Client> wclient; |
| status_t lStatus; |
| RecordThread *thread; |
| size_t inFrameCount; |
| int lSessionId; |
| |
| // check calling permissions |
| if (!recordingAllowed()) { |
| lStatus = PERMISSION_DENIED; |
| goto Exit; |
| } |
| |
| // add client to list |
| { // scope for mLock |
| Mutex::Autolock _l(mLock); |
| thread = checkRecordThread_l(input); |
| if (thread == NULL) { |
| lStatus = BAD_VALUE; |
| goto Exit; |
| } |
| |
| wclient = mClients.valueFor(pid); |
| if (wclient != NULL) { |
| client = wclient.promote(); |
| } else { |
| client = new Client(this, pid); |
| mClients.add(pid, client); |
| } |
| |
| // 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; |
| } |
| } |
| // create new record track. The record track uses one track in mHardwareMixerThread by convention. |
| recordTrack = thread->createRecordTrack_l(client, |
| sampleRate, |
| format, |
| channelMask, |
| frameCount, |
| flags, |
| lSessionId, |
| &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<AudioFlinger::RecordThread::RecordTrack>& recordTrack) |
| : BnAudioRecord(), |
| mRecordTrack(recordTrack) |
| { |
| } |
| |
| AudioFlinger::RecordHandle::~RecordHandle() { |
| stop(); |
| } |
| |
| status_t AudioFlinger::RecordHandle::start() { |
| ALOGV("RecordHandle::start()"); |
| return mRecordTrack->start(); |
| } |
| |
| void AudioFlinger::RecordHandle::stop() { |
| ALOGV("RecordHandle::stop()"); |
| mRecordTrack->stop(); |
| } |
| |
| sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { |
| return mRecordTrack->getCblk(); |
| } |
| |
| 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>& audioFlinger, |
| AudioStreamIn *input, |
| uint32_t sampleRate, |
| uint32_t channels, |
| int id, |
| uint32_t device) : |
| ThreadBase(audioFlinger, id, device), |
| mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL) |
| { |
| mType = ThreadBase::RECORD; |
| |
| snprintf(mName, kNameLength, "AudioIn_%d", id); |
| |
| mReqChannelCount = popcount(channels); |
| mReqSampleRate = sampleRate; |
| readInputParameters(); |
| } |
| |
| |
| AudioFlinger::RecordThread::~RecordThread() |
| { |
| delete[] mRsmpInBuffer; |
| if (mResampler != NULL) { |
| delete mResampler; |
| delete[] mRsmpOutBuffer; |
| } |
| } |
| |
| void AudioFlinger::RecordThread::onFirstRef() |
| { |
| run(mName, PRIORITY_URGENT_AUDIO); |
| } |
| |
| status_t AudioFlinger::RecordThread::readyToRun() |
| { |
| status_t status = initCheck(); |
| LOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); |
| return status; |
| } |
| |
| bool AudioFlinger::RecordThread::threadLoop() |
| { |
| AudioBufferProvider::Buffer buffer; |
| sp<RecordTrack> activeTrack; |
| Vector< sp<EffectChain> > effectChains; |
| |
| nsecs_t lastWarning = 0; |
| |
| acquireWakeLock(); |
| |
| // start recording |
| while (!exitPending()) { |
| |
| processConfigEvents(); |
| |
| { // scope for mLock |
| Mutex::Autolock _l(mLock); |
| checkForNewParameters_l(); |
| if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { |
| if (!mStandby) { |
| mInput->stream->common.standby(&mInput->stream->common); |
| mStandby = true; |
| } |
| |
| 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) { |
| if (!mStandby) { |
| mInput->stream->common.standby(&mInput->stream->common); |
| mStandby = true; |
| } |
| mActiveTrack.clear(); |
| mStartStopCond.broadcast(); |
| } else if (mActiveTrack->mState == TrackBase::RESUMING) { |
| if (mReqChannelCount != mActiveTrack->channelCount()) { |
| mActiveTrack.clear(); |
| mStartStopCond.broadcast(); |
| } else if (mBytesRead != 0) { |
| // 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; |
| } |
| } |
| 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)) { |
| 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 { |
| int16_t *src16 = (int16_t *)src; |
| int16_t *dst16 = (int16_t *)dst; |
| if (mChannelCount == 1) { |
| while (framesIn--) { |
| *dst16++ = *src16; |
| *dst16++ = *src16++; |
| } |
| } else { |
| while (framesIn--) { |
| *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); |
| src16 += 2; |
| } |
| } |
| } |
| } |
| if (framesOut && mFrameCount == mRsmpInIndex) { |
| if (framesOut == mFrameCount && |
| ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { |
| mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); |
| framesOut = 0; |
| } else { |
| mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); |
| mRsmpInIndex = 0; |
| } |
| if (mBytesRead < 0) { |
| LOGE("Error reading audio input"); |
| if (mActiveTrack->mState == TrackBase::ACTIVE) { |
| // Force input into standby so that it tries to |
| // recover at next read attempt |
| mInput->stream->common.standby(&mInput->stream->common); |
| 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); |
| // 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 |
| int16_t *src = (int16_t *)mRsmpOutBuffer; |
| int16_t *dst = buffer.i16; |
| while (framesOut--) { |
| *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); |
| src += 2; |
| } |
| } else { |
| ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); |
| } |
| |
| } |
| mActiveTrack->releaseBuffer(&buffer); |
| mActiveTrack->overflow(); |
| } |
| // client isn't retrieving buffers fast enough |
| else { |
| if (!mActiveTrack->setOverflow()) { |
| nsecs_t now = systemTime(); |
| if ((now - lastWarning) > kWarningThrottleNs) { |
| LOGW("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(); |
| } |
| |
| if (!mStandby) { |
| mInput->stream->common.standby(&mInput->stream->common); |
| } |
| mActiveTrack.clear(); |
| |
| mStartStopCond.broadcast(); |
| |
| releaseWakeLock(); |
| |
| ALOGV("RecordThread %p exiting", this); |
| return false; |
| } |
| |
| |
| sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( |
| const sp<AudioFlinger::Client>& client, |
| uint32_t sampleRate, |
| int format, |
| int channelMask, |
| int frameCount, |
| uint32_t flags, |
| int sessionId, |
| status_t *status) |
| { |
| sp<RecordTrack> track; |
| status_t lStatus; |
| |
| lStatus = initCheck(); |
| if (lStatus != NO_ERROR) { |
| LOGE("Audio driver not initialized."); |
| goto Exit; |
| } |
| |
| { // scope for mLock |
| Mutex::Autolock _l(mLock); |
| |
| track = new RecordTrack(this, client, sampleRate, |
| format, channelMask, frameCount, flags, sessionId); |
| |
| if (track->getCblk() == NULL) { |
| lStatus = NO_MEMORY; |
| goto Exit; |
| } |
| |
| mTrack = track.get(); |
| // disable AEC and NS if the device is a BT SCO headset supporting those pre processings |
| bool suspend = audio_is_bluetooth_sco_device( |
| (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && 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) |
| { |
| ALOGV("RecordThread::start"); |
| sp <ThreadBase> strongMe = this; |
| status_t status = NO_ERROR; |
| { |
| 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(); |
| return status; |
| } |
| mRsmpInIndex = mFrameCount; |
| mBytesRead = 0; |
| if (mResampler != NULL) { |
| mResampler->reset(); |
| } |
| mActiveTrack->mState = TrackBase::RESUMING; |
| // signal thread to start |
| ALOGV("Signal record thread"); |
| mWaitWorkCV.signal(); |
| // do not wait for mStartStopCond if exiting |
| if (mExiting) { |
| 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); |
| return status; |
| } |
| |
| void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { |
| ALOGV("RecordThread::stop"); |
| sp <ThreadBase> strongMe = this; |
| { |
| AutoMutex lock(&mLock); |
| if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { |
| mActiveTrack->mState = TrackBase::PAUSING; |
| // do not wait for mStartStopCond if exiting |
| if (mExiting) { |
| return; |
| } |
| mStartStopCond.wait(mLock); |
| // if we have been restarted, recordTrack == mActiveTrack.get() here |
| if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { |
| mLock.unlock(); |
| AudioSystem::stopInput(mId); |
| mLock.lock(); |
| ALOGV("Record stopped OK"); |
| } |
| } |
| } |
| } |
| |
| status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) |
| { |
| const size_t SIZE = 256; |
| char buffer[SIZE]; |
| String8 result; |
| pid_t pid = 0; |
| |
| snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); |
| result.append(buffer); |
| |
| if (mActiveTrack != 0) { |
| result.append("Active Track:\n"); |
| result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); |
| mActiveTrack->dump(buffer, SIZE); |
| result.append(buffer); |
| |
| 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 record client\n"); |
| } |
| write(fd, result.string(), result.size()); |
| |
| dumpBase(fd, args); |
| dumpEffectChains(fd, args); |
| |
| return NO_ERROR; |
| } |
| |
| status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) |
| { |
| 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) { |
| LOGE("RecordThread::getNextBuffer() Error reading audio input"); |
| if (mActiveTrack->mState == TrackBase::ACTIVE) { |
| // Force input into standby so that it tries to |
| // recover at next read attempt |
| mInput->stream->common.standby(&mInput->stream->common); |
| 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; |
| } |
| |
| 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; |
| int 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 = value; |
| reconfig = true; |
| } |
| if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { |
| reqChannelCount = popcount(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 garantied |
| // 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 (value & AUDIO_DEVICE_OUT_ALL) { |
| mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); |
| status = BAD_VALUE; |
| } else { |
| mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); |
| // disable AEC and NS if the device is a BT SCO headset supporting those pre processings |
| if (mTrack != NULL) { |
| bool suspend = audio_is_bluetooth_sco_device( |
| (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); |
| setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); |
| setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); |
| } |
| } |
| mDevice |= (uint32_t)value; |
| } |
| if (status == NO_ERROR) { |
| status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); |
| if (status == INVALID_OPERATION) { |
| mInput->stream->common.standby(&mInput->stream->common); |
| 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)) && |
| (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && |
| (reqChannelCount < 3)) { |
| status = NO_ERROR; |
| } |
| if (status == NO_ERROR) { |
| readInputParameters(); |
| sendConfigEvent_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 = 0; |
| |
| 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() |
| { |
| if (mRsmpInBuffer) delete mRsmpInBuffer; |
| if (mRsmpOutBuffer) delete mRsmpOutBuffer; |
| if (mResampler) 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)popcount(mChannelMask); |
| mFormat = mInput->stream->common.get_format(&mInput->stream->common); |
| mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common); |
| mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); |
| mFrameCount = mInputBytes / mFrameSize; |
| mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; |
| |
| if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) |
| { |
| int channelCount; |
| // optmization: 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) |
| { |
| Mutex::Autolock _l(mLock); |
| uint32_t result = 0; |
| if (getEffectChain_l(sessionId) != 0) { |
| result = EFFECT_SESSION; |
| } |
| |
| if (mTrack != NULL && sessionId == mTrack->sessionId()) { |
| result |= TRACK_SESSION; |
| } |
| |
| return result; |
| } |
| |
| AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() |
| { |
| Mutex::Autolock _l(mLock); |
| return mTrack; |
| } |
| |
| AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() |
| { |
| Mutex::Autolock _l(mLock); |
| return mInput; |
| } |
| |
| 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() |
| { |
| if (mInput == NULL) { |
| return NULL; |
| } |
| return &mInput->stream->common; |
| } |
| |
| |
| // ---------------------------------------------------------------------------- |
| |
| int AudioFlinger::openOutput(uint32_t *pDevices, |
| uint32_t *pSamplingRate, |
| uint32_t *pFormat, |
| uint32_t *pChannels, |
| uint32_t *pLatencyMs, |
| uint32_t flags) |
| { |
| status_t status; |
| PlaybackThread *thread = NULL; |
| mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; |
| uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; |
| uint32_t format = pFormat ? *pFormat : 0; |
| uint32_t channels = pChannels ? *pChannels : 0; |
| uint32_t latency = pLatencyMs ? *pLatencyMs : 0; |
| audio_stream_out_t *outStream; |
| audio_hw_device_t *outHwDev; |
| |
| ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", |
| pDevices ? *pDevices : 0, |
| samplingRate, |
| format, |
| channels, |
| flags); |
| |
| if (pDevices == NULL || *pDevices == 0) { |
| return 0; |
| } |
| |
| Mutex::Autolock _l(mLock); |
| |
| outHwDev = findSuitableHwDev_l(*pDevices); |
| if (outHwDev == NULL) |
| return 0; |
| |
| status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format, |
| &channels, &samplingRate, &outStream); |
| ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", |
| outStream, |
| samplingRate, |
| format, |
| channels, |
| status); |
| |
| mHardwareStatus = AUDIO_HW_IDLE; |
| if (outStream != NULL) { |
| AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); |
| int id = nextUniqueId(); |
| |
| if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || |
| (format != AUDIO_FORMAT_PCM_16_BIT) || |
| (channels != 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); |
| } |
| mPlaybackThreads.add(id, thread); |
| |
| if (pSamplingRate) *pSamplingRate = samplingRate; |
| if (pFormat) *pFormat = format; |
| if (pChannels) *pChannels = channels; |
| if (pLatencyMs) *pLatencyMs = thread->latency(); |
| |
| // notify client processes of the new output creation |
| thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); |
| return id; |
| } |
| |
| return 0; |
| } |
| |
| int AudioFlinger::openDuplicateOutput(int output1, int output2) |
| { |
| Mutex::Autolock _l(mLock); |
| MixerThread *thread1 = checkMixerThread_l(output1); |
| MixerThread *thread2 = checkMixerThread_l(output2); |
| |
| if (thread1 == NULL || thread2 == NULL) { |
| LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); |
| return 0; |
| } |
| |
| int 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(int output) |
| { |
| // keep strong reference on the playback thread so that |
| // it is not destroyed while exit() is executed |
| sp <PlaybackThread> 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)->type() == ThreadBase::DUPLICATING) { |
| DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); |
| dupThread->removeOutputTrack((MixerThread *)thread.get()); |
| } |
| } |
| } |
| void *param2 = 0; |
| audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); |
| mPlaybackThreads.removeItem(output); |
| } |
| thread->exit(); |
| |
| if (thread->type() != ThreadBase::DUPLICATING) { |
| AudioStreamOut *out = thread->clearOutput(); |
| // 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(int 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(int 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; |
| } |
| |
| int AudioFlinger::openInput(uint32_t *pDevices, |
| uint32_t *pSamplingRate, |
| uint32_t *pFormat, |
| uint32_t *pChannels, |
| uint32_t acoustics) |
| { |
| status_t status; |
| RecordThread *thread = NULL; |
| uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; |
| uint32_t format = pFormat ? *pFormat : 0; |
| uint32_t channels = pChannels ? *pChannels : 0; |
| uint32_t reqSamplingRate = samplingRate; |
| uint32_t reqFormat = format; |
| uint32_t reqChannels = channels; |
| audio_stream_in_t *inStream; |
| audio_hw_device_t *inHwDev; |
| |
| if (pDevices == NULL || *pDevices == 0) { |
| return 0; |
| } |
| |
| Mutex::Autolock _l(mLock); |
| |
| inHwDev = findSuitableHwDev_l(*pDevices); |
| if (inHwDev == NULL) |
| return 0; |
| |
| status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, |
| &channels, &samplingRate, |
| (audio_in_acoustics_t)acoustics, |
| &inStream); |
| ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", |
| inStream, |
| samplingRate, |
| format, |
| channels, |
| acoustics, |
| 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 (inStream == NULL && status == BAD_VALUE && |
| reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && |
| (samplingRate <= 2 * reqSamplingRate) && |
| (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { |
| ALOGV("openInput() reopening with proposed sampling rate and channels"); |
| status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, |
| &channels, &samplingRate, |
| (audio_in_acoustics_t)acoustics, |
| &inStream); |
| } |
| |
| if (inStream != NULL) { |
| AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); |
| |
| int id = nextUniqueId(); |
| // Start record thread |
| // RecorThread require both input and output device indication to forward to audio |
| // pre processing modules |
| uint32_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) *pSamplingRate = reqSamplingRate; |
| if (pFormat) *pFormat = format; |
| if (pChannels) *pChannels = reqChannels; |
| |
| input->stream->common.standby(&input->stream->common); |
| |
| // notify client processes of the new input creation |
| thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); |
| return id; |
| } |
| |
| return 0; |
| } |
| |
| status_t AudioFlinger::closeInput(int input) |
| { |
| // keep strong reference on the record thread so that |
| // it is not destroyed while exit() is executed |
| sp <RecordThread> thread; |
| { |
| Mutex::Autolock _l(mLock); |
| thread = checkRecordThread_l(input); |
| if (thread == NULL) { |
| return BAD_VALUE; |
| } |
| |
| ALOGV("closeInput() %d", input); |
| void *param2 = 0; |
| audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); |
| mRecordThreads.removeItem(input); |
| } |
| thread->exit(); |
| |
| AudioStreamIn *in = thread->clearInput(); |
| // 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(uint32_t stream, int output) |
| { |
| Mutex::Autolock _l(mLock); |
| MixerThread *dstThread = checkMixerThread_l(output); |
| if (dstThread == NULL) { |
| LOGW("setStreamOutput() bad output id %d", output); |
| return BAD_VALUE; |
| } |
| |
| ALOGV("setStreamOutput() stream %d to output %d", stream, output); |
| audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); |
| |
| dstThread->setStreamValid(stream, true); |
| |
| for (size_t i = 0; i < mPlaybackThreads.size(); i++) { |
| PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); |
| if (thread != dstThread && |
| thread->type() != ThreadBase::DIRECT) { |
| MixerThread *srcThread = (MixerThread *)thread; |
| srcThread->setStreamValid(stream, false); |
| srcThread->invalidateTracks(stream); |
| } |
| } |
| |
| return NO_ERROR; |
| } |
| |
| |
| int AudioFlinger::newAudioSessionId() |
| { |
| return nextUniqueId(); |
| } |
| |
| void AudioFlinger::acquireAudioSessionId(int audioSession) |
| { |
| Mutex::Autolock _l(mLock); |
| int caller = IPCThreadState::self()->getCallingPid(); |
| ALOGV("acquiring %d from %d", audioSession, caller); |
| int num = mAudioSessionRefs.size(); |
| for (int i = 0; i< num; i++) { |
| AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); |
| if (ref->sessionid == audioSession && ref->pid == caller) { |
| ref->cnt++; |
| ALOGV(" incremented refcount to %d", ref->cnt); |
| return; |
| } |
| } |
| AudioSessionRef *ref = new AudioSessionRef(); |
| ref->sessionid = audioSession; |
| ref->pid = caller; |
| ref->cnt = 1; |
| mAudioSessionRefs.push(ref); |
| ALOGV(" added new entry for %d", ref->sessionid); |
| } |
| |
| void AudioFlinger::releaseAudioSessionId(int audioSession) |
| { |
| Mutex::Autolock _l(mLock); |
| int caller = IPCThreadState::self()->getCallingPid(); |
| ALOGV("releasing %d from %d", audioSession, caller); |
| int num = mAudioSessionRefs.size(); |
| for (int i = 0; i< num; i++) { |
| AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); |
| if (ref->sessionid == audioSession && ref->pid == caller) { |
| ref->cnt--; |
| ALOGV(" decremented refcount to %d", ref->cnt); |
| if (ref->cnt == 0) { |
| mAudioSessionRefs.removeAt(i); |
| delete ref; |
| purgeStaleEffects_l(); |
| } |
| return; |
| } |
| } |
| LOGW("session id %d not found for pid %d", audioSession, caller); |
| } |
| |
| void AudioFlinger::purgeStaleEffects_l() { |
| |
| ALOGV("purging stale effects"); |
| |
| Vector< sp<EffectChain> > chains; |
| |
| for (size_t i = 0; i < mPlaybackThreads.size(); i++) { |
| sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); |
| for (size_t j = 0; j < t->mEffectChains.size(); j++) { |
| sp<EffectChain> ec = t->mEffectChains[j]; |
| if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { |
| chains.push(ec); |
| } |
| } |
| } |
| for (size_t i = 0; i < mRecordThreads.size(); i++) { |
| sp<RecordThread> t = mRecordThreads.valueAt(i); |
| for (size_t j = 0; j < t->mEffectChains.size(); j++) { |
| sp<EffectChain> ec = t->mEffectChains[j]; |
| chains.push(ec); |
| } |
| } |
| |
| for (size_t i = 0; i < chains.size(); i++) { |
| sp<EffectChain> ec = chains[i]; |
| int sessionid = ec->sessionId(); |
| sp<ThreadBase> 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->sessionid == sessionid) { |
| ALOGV(" session %d still exists for %d with %d refs", |
| sessionid, ref->pid, ref->cnt); |
| found = true; |
| break; |
| } |
| } |
| if (!found) { |
| // remove all effects from the chain |
| while (ec->mEffects.size()) { |
| sp<EffectModule> effect = ec->mEffects[0]; |
| effect->unPin(); |
| Mutex::Autolock _l (t->mLock); |
| t->removeEffect_l(effect); |
| for (size_t j = 0; j < effect->mHandles.size(); j++) { |
| sp<EffectHandle> handle = effect->mHandles[j].promote(); |
| if (handle != 0) { |
| handle->mEffect.clear(); |
| if (handle->mHasControl && handle->mEnabled) { |
| 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(int output) const |
| { |
| PlaybackThread *thread = NULL; |
| if (mPlaybackThreads.indexOfKey(output) >= 0) { |
| thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); |
| } |
| return thread; |
| } |
| |
| // checkMixerThread_l() must be called with AudioFlinger::mLock held |
| AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const |
| { |
| PlaybackThread *thread = checkPlaybackThread_l(output); |
| if (thread != NULL) { |
| if (thread->type() == ThreadBase::DIRECT) { |
| thread = NULL; |
| } |
| } |
| return (MixerThread *)thread; |
| } |
| |
| // checkRecordThread_l() must be called with AudioFlinger::mLock held |
| AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const |
| { |
| RecordThread *thread = NULL; |
| if (mRecordThreads.indexOfKey(input) >= 0) { |
| thread = (RecordThread *)mRecordThreads.valueFor(input).get(); |
| } |
| return thread; |
| } |
| |
| uint32_t AudioFlinger::nextUniqueId() |
| { |
| return android_atomic_inc(&mNextUniqueId); |
| } |
| |
| AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() |
| { |
| for (size_t i = 0; i < mPlaybackThreads.size(); i++) { |
| PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); |
| AudioStreamOut *output = thread->getOutput(); |
| if (output != NULL && output->hwDev == mPrimaryHardwareDev) { |
| return thread; |
| } |
| } |
| return NULL; |
| } |
| |
| uint32_t AudioFlinger::primaryOutputDevice_l() |
| { |
| PlaybackThread *thread = primaryPlaybackThread_l(); |
| |
| if (thread == NULL) { |
| return 0; |
| } |
| |
| return thread->device(); |
| } |
| |
| |
| // ---------------------------------------------------------------------------- |
| // Effect management |
| // ---------------------------------------------------------------------------- |
| |
| |
| status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) |
| { |
| Mutex::Autolock _l(mLock); |
| return EffectQueryNumberEffects(numEffects); |
| } |
| |
| status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) |
| { |
| Mutex::Autolock _l(mLock); |
| return EffectQueryEffect(index, descriptor); |
| } |
| |
| status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) |
| { |
| Mutex::Autolock _l(mLock); |
| return EffectGetDescriptor(pUuid, descriptor); |
| } |
| |
| |
| sp<IEffect> AudioFlinger::createEffect(pid_t pid, |
| effect_descriptor_t *pDesc, |
| const sp<IEffectClient>& effectClient, |
| int32_t priority, |
| int io, |
| int sessionId, |
| status_t *status, |
| int *id, |
| int *enabled) |
| { |
| status_t lStatus = NO_ERROR; |
| sp<EffectHandle> handle; |
| effect_descriptor_t desc; |
| sp<Client> client; |
| wp<Client> wclient; |
| |
| ALOGV("createEffect pid %d, client %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() != 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) { |
| LOGW("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)) { |
| LOGW("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) { |
| LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); |
| goto Exit; |
| } |
| for (uint32_t i = 0; i < numEffects; i++) { |
| lStatus = EffectQueryEffect(i, &desc); |
| if (lStatus < 0) { |
| LOGW("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; |
| memcpy(&d, &desc, sizeof(effect_descriptor_t)); |
| if (sessionId != AUDIO_SESSION_OUTPUT_MIX || |
| (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { |
| break; |
| } |
| } |
| } |
| if (!found) { |
| lStatus = BAD_VALUE; |
| LOGW("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) { |
| memcpy(&desc, &d, sizeof(effect_descriptor_t)); |
| } |
| } |
| |
| // 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 |
| memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); |
| |
| // 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) { |
| LOGE("createEffect() unknown output thread"); |
| lStatus = BAD_VALUE; |
| goto Exit; |
| } |
| } |
| |
| wclient = mClients.valueFor(pid); |
| |
| if (wclient != NULL) { |
| client = wclient.promote(); |
| } else { |
| client = new Client(this, pid); |
| mClients.add(pid, client); |
| } |
| |
| // 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) { |
| *status = lStatus; |
| } |
| return handle; |
| } |
| |
| status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) |
| { |
| ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", |
| sessionId, srcOutput, dstOutput); |
| Mutex::Autolock _l(mLock); |
| if (srcOutput == dstOutput) { |
| LOGW("moveEffects() same dst and src outputs %d", dstOutput); |
| return NO_ERROR; |
| } |
| PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); |
| if (srcThread == NULL) { |
| LOGW("moveEffects() bad srcOutput %d", srcOutput); |
| return BAD_VALUE; |
| } |
| PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); |
| if (dstThread == NULL) { |
| LOGW("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<EffectChain> chain = srcThread->getEffectChain_l(sessionId); |
| if (chain == 0) { |
| LOGW("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 |
| int dstOutput = dstThread->id(); |
| sp<EffectChain> dstChain; |
| uint32_t strategy = 0; // prevent compiler warning |
| sp<EffectModule> 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) { |
| LOGW("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::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( |
| const sp<AudioFlinger::Client>& client, |
| const sp<IEffectClient>& effectClient, |
| int32_t priority, |
| int sessionId, |
| effect_descriptor_t *desc, |
| int *enabled, |
| status_t *status |
| ) |
| { |
| sp<EffectModule> effect; |
| sp<EffectHandle> handle; |
| status_t lStatus; |
| sp<EffectChain> chain; |
| bool chainCreated = false; |
| bool effectCreated = false; |
| bool effectRegistered = false; |
| |
| lStatus = initCheck(); |
| if (lStatus != NO_ERROR) { |
| LOGW("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) { |
| LOGW("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) || |
| (mType != RECORD && |
| (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { |
| LOGW("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; |
| } else { |
| effect = chain->getEffectFromDesc_l(desc); |
| } |
| |
| ALOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : 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(mDevice); |
| effect->setMode(mAudioFlinger->getMode()); |
| } |
| // create effect handle and connect it to effect module |
| handle = new EffectHandle(effect, client, effectClient, priority); |
| lStatus = effect->addHandle(handle); |
| if (enabled) { |
| *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) { |
| *status = lStatus; |
| } |
| return handle; |
| } |
| |
| sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) |
| { |
| sp<EffectModule> effect; |
| |
| sp<EffectChain> chain = getEffectChain_l(sessionId); |
| if (chain != 0) { |
| effect = chain->getEffectFromId_l(effectId); |
| } |
| return effect; |
| } |
| |
| // PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and |
| // PlaybackThread::mLock held |
| status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) |
| { |
| // check for existing effect chain with the requested audio session |
| int sessionId = effect->sessionId(); |
| sp<EffectChain> chain = getEffectChain_l(sessionId); |
| bool chainCreated = false; |
| |
| 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) { |
| LOGW("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(mDevice); |
| effect->setMode(mAudioFlinger->getMode()); |
| return NO_ERROR; |
| } |
| |
| void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { |
| |
| ALOGV("removeEffect_l() %p effect %p", this, effect.get()); |
| effect_descriptor_t desc = effect->desc(); |
| if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { |
| detachAuxEffect_l(effect->id()); |
| } |
| |
| sp<EffectChain> chain = effect->chain().promote(); |
| if (chain != 0) { |
| // remove effect chain if removing last effect |
| if (chain->removeEffect_l(effect) == 0) { |
| removeEffectChain_l(chain); |
| } |
| } else { |
| LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); |
| } |
| } |
| |
| void AudioFlinger::ThreadBase::lockEffectChains_l( |
| Vector<sp <AudioFlinger::EffectChain> >& effectChains) |
| { |
| effectChains = mEffectChains; |
| for (size_t i = 0; i < mEffectChains.size(); i++) { |
| mEffectChains[i]->lock(); |
| } |
| } |
| |
| void AudioFlinger::ThreadBase::unlockEffectChains( |
| Vector<sp <AudioFlinger::EffectChain> >& effectChains) |
| { |
| for (size_t i = 0; i < effectChains.size(); i++) { |
| effectChains[i]->unlock(); |
| } |
| } |
| |
| sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) |
| { |
| Mutex::Autolock _l(mLock); |
| return getEffectChain_l(sessionId); |
| } |
| |
| sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) |
| { |
| sp<EffectChain> chain; |
| |
| size_t size = mEffectChains.size(); |
| for (size_t i = 0; i < size; i++) { |
| if (mEffectChains[i]->sessionId() == sessionId) { |
| chain = mEffectChains[i]; |
| break; |
| } |
| } |
| return chain; |
| } |
| |
| void AudioFlinger::ThreadBase::setMode(uint32_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<EffectModule>& effect, |
| const wp<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<EffectChain>& 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 = mFrameCount * mChannelCount; |
| buffer = new int16_t[numSamples]; |
| memset(buffer, 0, numSamples * sizeof(int16_t)); |
| ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); |
| ownsBuffer = true; |
| } |
| |
| // Attach all tracks with same session ID to this chain. |
| for (size_t i = 0; i < mTracks.size(); ++i) { |
| sp<Track> track = mTracks[i]; |
| if (session == track->sessionId()) { |
| ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); |
| track->setMainBuffer(buffer); |
| chain->incTrackCnt(); |
| } |
| } |
| |
| // indicate all active tracks in the chain |
| for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { |
| sp<Track> track = mActiveTracks[i].promote(); |
| if (track == 0) continue; |
| if (session == track->sessionId()) { |
| ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); |
| chain->incActiveTrackCnt(); |
| } |
| } |
| } |
| |
| chain->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<EffectChain>& chain) |
| { |
| int session = chain->sessionId(); |
| |
| ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); |
| |
| for (size_t i = 0; i < mEffectChains.size(); i++) { |
| if (chain == mEffectChains[i]) { |
| mEffectChains.removeAt(i); |
| // detach all active tracks from the chain |
| for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { |
| sp<Track> track = mActiveTracks[i].promote(); |
| if (track == 0) continue; |
| if (session == track->sessionId()) { |
| ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", |
| chain.get(), session); |
| chain->decActiveTrackCnt(); |
| } |
| } |
| |
| // detach all tracks with same session ID from this chain |
| for (size_t i = 0; i < mTracks.size(); ++i) { |
| sp<Track> track = mTracks[i]; |
| if (session == track->sessionId()) { |
| track->setMainBuffer(mMixBuffer); |
| chain->decTrackCnt(); |
| } |
| } |
| break; |
| } |
| } |
| return mEffectChains.size(); |
| } |
| |
| status_t AudioFlinger::PlaybackThread::attachAuxEffect( |
| const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) |
| { |
| Mutex::Autolock _l(mLock); |
| return attachAuxEffect_l(track, EffectId); |
| } |
| |
| status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( |
| const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) |
| { |
| status_t status = NO_ERROR; |
| |
| if (EffectId == 0) { |
| track->setAuxBuffer(0, NULL); |
| } else { |
| // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX |
| sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); |
| if (effect != 0) { |
| if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { |
| track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); |
| } else { |
| status = INVALID_OPERATION; |
| } |
| } else { |
| status = BAD_VALUE; |
| } |
| } |
| return status; |
| } |
| |
| void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) |
| { |
| for (size_t i = 0; i < mTracks.size(); ++i) { |
| sp<Track> track = mTracks[i]; |
| if (track->auxEffectId() == effectId) { |
| attachAuxEffect_l(track, 0); |
| } |
| } |
| } |
| |
| status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& 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<EffectChain>& chain) |
| { |
| ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); |
| LOGW_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(const wp<ThreadBase>& wThread, |
| const wp<AudioFlinger::EffectChain>& chain, |
| effect_descriptor_t *desc, |
| int id, |
| int sessionId) |
| : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), |
| mStatus(NO_INIT), mState(IDLE), mSuspended(false) |
| { |
| ALOGV("Constructor %p", this); |
| int lStatus; |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread == 0) { |
| return; |
| } |
| |
| memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); |
| |
| // 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; |
| } |
| |
| if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { |
| mPinned = true; |
| } |
| 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<ThreadBase> 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(sp<EffectHandle>& handle) |
| { |
| status_t status; |
| |
| Mutex::Autolock _l(mLock); |
| // First handle in mHandles has highest priority and controls the effect module |
| int priority = handle->priority(); |
| size_t size = mHandles.size(); |
| sp<EffectHandle> h; |
| size_t i; |
| for (i = 0; i < size; i++) { |
| h = mHandles[i].promote(); |
| if (h == 0) continue; |
| 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 (h != 0) { |
| enabled = h->enabled(); |
| h->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.get(), i); |
| mHandles.insertAt(handle, i); |
| return status; |
| } |
| |
| size_t AudioFlinger::EffectModule::removeHandle(const wp<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.unsafe_get(), i); |
| |
| bool enabled = false; |
| EffectHandle *hdl = handle.unsafe_get(); |
| if (hdl) { |
| ALOGV("removeHandle() unsafe_get OK"); |
| enabled = hdl->enabled(); |
| } |
| mHandles.removeAt(i); |
| size = mHandles.size(); |
| // if removed from first place, move effect control from this handle to next in line |
| if (i == 0 && size != 0) { |
| sp<EffectHandle> h = mHandles[0].promote(); |
| if (h != 0) { |
| h->setControl(true /*hasControl*/, true /*signal*/ , 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 (size == 0 && !mPinned) { |
| mState = DESTROYED; |
| } |
| |
| return size; |
| } |
| |
| sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() |
| { |
| Mutex::Autolock _l(mLock); |
| sp<EffectHandle> handle; |
| if (mHandles.size() != 0) { |
| handle = mHandles[0].promote(); |
| } |
| return handle; |
| } |
| |
| void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) |
| { |
| ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); |
| // keep a strong reference on this EffectModule to avoid calling the |
| // destructor before we exit |
| sp<EffectModule> keep(this); |
| { |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| thread->disconnectEffect(keep, handle, unpiniflast); |
| } |
| } |
| } |
| |
| 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<EffectChain> 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); |
| } |
| |
| status_t AudioFlinger::EffectModule::configure() |
| { |
| uint32_t channels; |
| if (mEffectInterface == NULL) { |
| return NO_INIT; |
| } |
| |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread == 0) { |
| return DEAD_OBJECT; |
| } |
| |
| // TODO: handle configuration of effects replacing track process |
| if (thread->channelCount() == 1) { |
| channels = AUDIO_CHANNEL_OUT_MONO; |
| } else { |
| channels = AUDIO_CHANNEL_OUT_STEREO; |
| } |
| |
| if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { |
| mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; |
| } else { |
| mConfig.inputCfg.channels = channels; |
| } |
| mConfig.outputCfg.channels = channels; |
| mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; |
| mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; |
| 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; |
| 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; |
| } |
| |
| 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<ThreadBase> 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<ThreadBase> 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++) { |
| sp<EffectHandle> h = mHandles[i].promote(); |
| if (h != 0) { |
| h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); |
| } |
| } |
| } |
| return status; |
| } |
| |
| status_t AudioFlinger::EffectModule::setEnabled(bool enabled) |
| { |
| |
| Mutex::Autolock _l(mLock); |
| ALOGV("setEnabled %p enabled %d", this, enabled); |
| |
| if (enabled != isEnabled()) { |
| 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++) { |
| sp<EffectHandle> h = mHandles[i].promote(); |
| if (h != 0) { |
| h->setEnabled(enabled); |
| } |
| } |
| } |
| return NO_ERROR; |
| } |
| |
| bool AudioFlinger::EffectModule::isEnabled() |
| { |
| 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() |
| { |
| 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(uint32_t device) |
| { |
| Mutex::Autolock _l(mLock); |
| status_t status = NO_ERROR; |
| if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { |
| // audio pre processing modules on RecordThread can receive both output and |
| // input device indication in the same call |
| uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; |
| if (dev) { |
| status_t cmdStatus; |
| uint32_t size = sizeof(status_t); |
| |
| status = (*mEffectInterface)->command(mEffectInterface, |
| EFFECT_CMD_SET_DEVICE, |
| sizeof(uint32_t), |
| &dev, |
| &size, |
| &cmdStatus); |
| if (status == NO_ERROR) { |
| status = cmdStatus; |
| } |
| } |
| dev = device & AUDIO_DEVICE_IN_ALL; |
| if (dev) { |
| status_t cmdStatus; |
| uint32_t size = sizeof(status_t); |
| |
| status_t status2 = (*mEffectInterface)->command(mEffectInterface, |
| EFFECT_CMD_SET_INPUT_DEVICE, |
| sizeof(uint32_t), |
| &dev, |
| &size, |
| &cmdStatus); |
| if (status2 == NO_ERROR) { |
| status2 = cmdStatus; |
| } |
| if (status == NO_ERROR) { |
| status = status2; |
| } |
| } |
| } |
| return status; |
| } |
| |
| status_t AudioFlinger::EffectModule::setMode(uint32_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(int), |
| &mode, |
| &size, |
| &cmdStatus); |
| if (status == NO_ERROR) { |
| status = cmdStatus; |
| } |
| } |
| return status; |
| } |
| |
| void AudioFlinger::EffectModule::setSuspended(bool suspended) |
| { |
| Mutex::Autolock _l(mLock); |
| mSuspended = suspended; |
| } |
| bool AudioFlinger::EffectModule::suspended() |
| { |
| Mutex::Autolock _l(mLock); |
| return mSuspended; |
| } |
| |
| status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& 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) { |
| sp<EffectHandle> handle = mHandles[i].promote(); |
| if (handle != 0) { |
| handle->dump(buffer, SIZE); |
| result.append(buffer); |
| } |
| } |
| |
| result.append("\n"); |
| |
| write(fd, result.string(), result.length()); |
| |
| if (locked) { |
| mLock.unlock(); |
| } |
| |
| return NO_ERROR; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // EffectHandle implementation |
| // ---------------------------------------------------------------------------- |
| |
| #undef LOG_TAG |
| #define LOG_TAG "AudioFlinger::EffectHandle" |
| |
| AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, |
| const sp<AudioFlinger::Client>& client, |
| const sp<IEffectClient>& effectClient, |
| int32_t priority) |
| : BnEffect(), |
| mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), |
| mPriority(priority), mHasControl(false), mEnabled(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<effect_param_cblk_t *>(mCblkMemory->pointer()); |
| |
| if (mCblk) { |
| new(mCblk) effect_param_cblk_t(); |
| mBuffer = (uint8_t *)mCblk + bufOffset; |
| } |
| } else { |
| LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); |
| return; |
| } |
| } |
| |
| AudioFlinger::EffectHandle::~EffectHandle() |
| { |
| ALOGV("Destructor %p", this); |
| disconnect(false); |
| ALOGV("Destructor DONE %p", this); |
| } |
| |
| 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<ThreadBase> 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<ThreadBase> 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; |
| } |
| mEffect->disconnect(this, unpiniflast); |
| |
| if (mHasControl && mEnabled) { |
| sp<ThreadBase> 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) { |
| mCblk->~effect_param_cblk_t(); // destroy our shared-structure. |
| } |
| mCblkMemory.clear(); // and free the shared memory |
| 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) { |
| LOGW("command(): invalid parameter block size"); |
| break; |
| } |
| effect_param_t *param = (effect_param_t *)p; |
| if (param->psize == 0 || param->vsize == 0) { |
| LOGW("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(); |
| } |
| |
| return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); |
| } |
| |
| sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { |
| return mCblkMemory; |
| } |
| |
| 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 ? tryLock(mCblk->lock) : false; |
| |
| snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", |
| (mClient == NULL) ? getpid() : 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(const wp<ThreadBase>& wThread, |
| int sessionId) |
| : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), |
| mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), |
| mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) |
| { |
| mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread == 0) { |
| 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::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) |
| { |
| sp<EffectModule> effect; |
| 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) { |
| effect = mEffects[i]; |
| break; |
| } |
| } |
| return effect; |
| } |
| |
| // getEffectFromId_l() must be called with ThreadBase::mLock held |
| sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) |
| { |
| sp<EffectModule> effect; |
| 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) { |
| effect = mEffects[i]; |
| break; |
| } |
| } |
| return effect; |
| } |
| |
| // getEffectFromType_l() must be called with ThreadBase::mLock held |
| sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( |
| const effect_uuid_t *type) |
| { |
| sp<EffectModule> effect; |
| 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) { |
| effect = mEffects[i]; |
| break; |
| } |
| } |
| return effect; |
| } |
| |
| // Must be called with EffectChain::mLock locked |
| void AudioFlinger::EffectChain::process_l() |
| { |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread == 0) { |
| LOGW("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) { |
| size_t numSamples = thread->frameCount() * thread->channelCount(); |
| memset(mInBuffer, 0, numSamples * sizeof(int16_t)); |
| if (mTailBufferCount > 0) { |
| mTailBufferCount--; |
| } |
| } |
| } |
| } |
| |
| size_t size = mEffects.size(); |
| if (doProcess) { |
| 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<EffectModule>& effect) |
| { |
| effect_descriptor_t desc = effect->desc(); |
| uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; |
| |
| Mutex::Autolock _l(mLock); |
| effect->setChain(this); |
| sp<ThreadBase> 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 |
| |
| int size = (int)mEffects.size(); |
| int idx_insert = size; |
| int idx_insert_first = -1; |
| int idx_insert_last = -1; |
| |
| for (int 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) { |
| LOGW("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<EffectModule>& effect) |
| { |
| Mutex::Autolock _l(mLock); |
| int size = (int)mEffects.size(); |
| int i; |
| uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; |
| |
| for (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(uint32_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(uint32_t mode) |
| { |
| size_t size = mEffects.size(); |
| for (size_t i = 0; i < size; i++) { |
| mEffects[i]->setMode(mode); |
| } |
| } |
| |
| // 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; |
| } |
| |
| status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& 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<EffectModule> effect = mEffects[i]; |
| if (effect != 0) { |
| effect->dump(fd, args); |
| } |
| } |
| |
| if (locked) { |
| mLock.unlock(); |
| } |
| |
| return NO_ERROR; |
| } |
| |
| // must be called with ThreadBase::mLock held |
| void AudioFlinger::EffectChain::setEffectSuspended_l( |
| const effect_uuid_t *type, bool suspend) |
| { |
| sp<SuspendedEffectDesc> desc; |
| // use effect type UUID timelow as key as there is no real risk of identical |
| // timeLow fields among effect type UUIDs. |
| int index = mSuspendedEffects.indexOfKey(type->timeLow); |
| if (suspend) { |
| if (index >= 0) { |
| desc = mSuspendedEffects.valueAt(index); |
| } else { |
| desc = new SuspendedEffectDesc(); |
| memcpy(&desc->mType, type, sizeof(effect_uuid_t)); |
| mSuspendedEffects.add(type->timeLow, desc); |
| ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); |
| } |
| if (desc->mRefCount++ == 0) { |
| sp<EffectModule> 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) { |
| LOGW("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<EffectModule> effect = desc->mEffect.promote(); |
| if (effect != 0) { |
| effect->setSuspended(false); |
| sp<EffectHandle> handle = effect->controlHandle(); |
| if (handle != 0) { |
| effect->setEnabled(handle->enabled()); |
| } |
| } |
| desc->mEffect.clear(); |
| } |
| mSuspendedEffects.removeItemsAt(index); |
| } |
| } |
| } |
| |
| // must be called with ThreadBase::mLock held |
| void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) |
| { |
| sp<SuspendedEffectDesc> desc; |
| |
| int 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<EffectModule> > effects = getSuspendEligibleEffects(); |
| 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) { |
| LOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); |
| desc->mRefCount = 1; |
| } |
| if (--desc->mRefCount == 0) { |
| Vector<const effect_uuid_t *> 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; |
| } |
| |
| Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects() |
| { |
| Vector< sp<EffectModule> > effects; |
| for (size_t i = 0; i < mEffects.size(); i++) { |
| if (!isEffectEligibleForSuspend(mEffects[i]->desc())) { |
| continue; |
| } |
| effects.add(mEffects[i]); |
| } |
| return effects; |
| } |
| |
| sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( |
| const effect_uuid_t *type) |
| { |
| sp<EffectModule> effect; |
| effect = getEffectFromType_l(type); |
| if (effect != 0 && !effect->isEnabled()) { |
| effect.clear(); |
| } |
| return effect; |
| } |
| |
| void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, |
| bool enabled) |
| { |
| int 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) { |
| LOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); |
| return; |
| } |
| } |
| ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", |
| effect->desc().type.timeLow); |
| sp<SuspendedEffectDesc> 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<SuspendedEffectDesc> 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 |