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-rw-r--r--services/audioflinger/AudioFlinger.cpp6419
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diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
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+++ b/services/audioflinger/AudioFlinger.cpp
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+/* //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/IServiceManager.h>
+#include <utils/Log.h>
+#include <binder/Parcel.h>
+#include <binder/IPCThreadState.h>
+#include <utils/String16.h>
+#include <utils/threads.h>
+
+#include <cutils/properties.h>
+
+#include <media/AudioTrack.h>
+#include <media/AudioRecord.h>
+
+#include <private/media/AudioTrackShared.h>
+#include <private/media/AudioEffectShared.h>
+#include <hardware_legacy/AudioHardwareInterface.h>
+
+#include "AudioMixer.h"
+#include "AudioFlinger.h"
+
+#ifdef WITH_A2DP
+#include "A2dpAudioInterface.h"
+#endif
+
+#ifdef LVMX
+#include "lifevibes.h"
+#endif
+
+#include <media/EffectsFactoryApi.h>
+#include <media/EffectVisualizerApi.h>
+
+// ----------------------------------------------------------------------------
+// the sim build doesn't have gettid
+
+#ifndef HAVE_GETTID
+# define gettid getpid
+#endif
+
+// ----------------------------------------------------------------------------
+
+extern const char * const gEffectLibPath;
+
+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 kDumpLockSleep = 20000;
+
+static const nsecs_t kWarningThrottle = seconds(5);
+
+
+#define AUDIOFLINGER_SECURITY_ENABLED 1
+
+// ----------------------------------------------------------------------------
+
+static bool recordingAllowed() {
+#ifndef HAVE_ANDROID_OS
+ return true;
+#endif
+#if AUDIOFLINGER_SECURITY_ENABLED
+ 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;
+#else
+ if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO")))
+ LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest");
+ return true;
+#endif
+}
+
+static bool settingsAllowed() {
+#ifndef HAVE_ANDROID_OS
+ return true;
+#endif
+#if AUDIOFLINGER_SECURITY_ENABLED
+ 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;
+#else
+ if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")))
+ LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest");
+ return true;
+#endif
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::AudioFlinger()
+ : BnAudioFlinger(),
+ mAudioHardware(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1)
+{
+ mHardwareStatus = AUDIO_HW_IDLE;
+
+ mAudioHardware = AudioHardwareInterface::create();
+
+ mHardwareStatus = AUDIO_HW_INIT;
+ if (mAudioHardware->initCheck() == NO_ERROR) {
+ // open 16-bit output stream for s/w mixer
+ mMode = AudioSystem::MODE_NORMAL;
+ setMode(mMode);
+
+ setMasterVolume(1.0f);
+ setMasterMute(false);
+ } else {
+ LOGE("Couldn't even initialize the stubbed audio hardware!");
+ }
+#ifdef LVMX
+ LifeVibes::init();
+ mLifeVibesClientPid = -1;
+#endif
+}
+
+AudioFlinger::~AudioFlinger()
+{
+ 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));
+ }
+ if (mAudioHardware) {
+ delete mAudioHardware;
+ }
+}
+
+
+
+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);
+ }
+ }
+ }
+ 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(kDumpLockSleep);
+ }
+ 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);
+ }
+
+ if (mAudioHardware) {
+ mAudioHardware->dumpState(fd, args);
+ }
+ if (locked) mLock.unlock();
+ }
+ return NO_ERROR;
+}
+
+
+// IAudioFlinger interface
+
+
+sp<IAudioTrack> AudioFlinger::createTrack(
+ pid_t pid,
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ 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 >= AudioSystem::NUM_STREAM_TYPES) {
+ LOGE("invalid stream type");
+ 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);
+ }
+
+ LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
+ if (sessionId != NULL && *sessionId != AudioSystem::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) {
+ 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;
+ }
+ }
+ LOGV("createTrack() lSessionId: %d", lSessionId);
+
+ track = thread->createTrack_l(client, streamType, sampleRate, format,
+ channelCount, 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();
+}
+
+int 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)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+
+ // when hw supports master volume, don't scale in sw mixer
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+ if (mAudioHardware->setMasterVolume(value) == NO_ERROR) {
+ value = 1.0f;
+ }
+ mHardwareStatus = AUDIO_HW_IDLE;
+
+ 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;
+
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+ if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) {
+ LOGW("Illegal value: setMode(%d)", mode);
+ return BAD_VALUE;
+ }
+
+ { // scope for the lock
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_HW_SET_MODE;
+ ret = mAudioHardware->setMode(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);
+#ifdef LVMX
+ LifeVibes::setMode(mode);
+#endif
+ }
+
+ return ret;
+}
+
+status_t AudioFlinger::setMicMute(bool state)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
+ status_t ret = mAudioHardware->setMicMute(state);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return ret;
+}
+
+bool AudioFlinger::getMicMute() const
+{
+ bool state = AudioSystem::MODE_INVALID;
+ mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
+ mAudioHardware->getMicMute(&state);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return state;
+}
+
+status_t AudioFlinger::setMasterMute(bool muted)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+
+ 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) >= AudioSystem::NUM_STREAM_TYPES) {
+ 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) >= AudioSystem::NUM_STREAM_TYPES ||
+ uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) {
+ return BAD_VALUE;
+ }
+
+ 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) >= AudioSystem::NUM_STREAM_TYPES) {
+ 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)AudioSystem::NUM_STREAM_TYPES) {
+ return true;
+ }
+
+ return mStreamTypes[stream].mute;
+}
+
+bool AudioFlinger::isStreamActive(int stream) const
+{
+ Mutex::Autolock _l(mLock);
+ for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
+ if (mPlaybackThreads.valueAt(i)->isStreamActive(stream)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
+{
+ status_t result;
+
+ LOGV("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;
+ }
+
+#ifdef LVMX
+ AudioParameter param = AudioParameter(keyValuePairs);
+ LifeVibes::setParameters(ioHandle,keyValuePairs);
+ String8 key = String8(AudioParameter::keyRouting);
+ int device;
+ if (NO_ERROR != param.getInt(key, device)) {
+ device = -1;
+ }
+
+ key = String8(LifevibesTag);
+ String8 value;
+ int musicEnabled = -1;
+ if (NO_ERROR == param.get(key, value)) {
+ if (value == LifevibesEnable) {
+ mLifeVibesClientPid = IPCThreadState::self()->getCallingPid();
+ musicEnabled = 1;
+ } else if (value == LifevibesDisable) {
+ mLifeVibesClientPid = -1;
+ musicEnabled = 0;
+ }
+ }
+#endif
+
+ // ioHandle == 0 means the parameters are global to the audio hardware interface
+ if (ioHandle == 0) {
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_SET_PARAMETER;
+ result = mAudioHardware->setParameters(keyValuePairs);
+#ifdef LVMX
+ if (musicEnabled != -1) {
+ LifeVibes::enableMusic((bool) musicEnabled);
+ }
+#endif
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return 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);
+ }
+ }
+ if (thread != NULL) {
+ result = thread->setParameters(keyValuePairs);
+#ifdef LVMX
+ if ((NO_ERROR == result) && (device != -1)) {
+ LifeVibes::setDevice(LifeVibes::threadIdToAudioOutputType(thread->id()), device);
+ }
+#endif
+ return result;
+ }
+ return BAD_VALUE;
+}
+
+String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
+{
+// LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
+// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
+
+ if (ioHandle == 0) {
+ return mAudioHardware->getParameters(keys);
+ }
+
+ 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)
+{
+ return mAudioHardware->getInputBufferSize(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)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
+ status_t ret = mAudioHardware->setVoiceVolume(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);
+ LOGV("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);
+ LOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
+#ifdef LVMX
+ if (pid == mLifeVibesClientPid) {
+ LOGV("Disabling lifevibes");
+ LifeVibes::enableMusic(false);
+ mLifeVibesClientPid = -1;
+ }
+#endif
+ mNotificationClients.removeItem(pid);
+ }
+}
+
+// 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)
+{
+ LOGV("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)
+ : Thread(false),
+ mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
+ mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false)
+{
+}
+
+AudioFlinger::ThreadBase::~ThreadBase()
+{
+ mParamCond.broadcast();
+ mNewParameters.clear();
+}
+
+void AudioFlinger::ThreadBase::exit()
+{
+ // keep a strong ref on ourself so that we wont get
+ // destroyed in the middle of requestExitAndWait()
+ sp <ThreadBase> strongMe = this;
+
+ LOGV("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;
+}
+
+int 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;
+
+ LOGV("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, seconds(2)) == 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 = new ConfigEvent();
+ configEvent->mEvent = event;
+ configEvent->mParam = param;
+ mConfigEvents.add(configEvent);
+ LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
+ mWaitWorkCV.signal();
+}
+
+void AudioFlinger::ThreadBase::processConfigEvents()
+{
+ mLock.lock();
+ while(!mConfigEvents.isEmpty()) {
+ LOGV("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();
+ delete configEvent;
+ 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, "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;
+}
+
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
+ : ThreadBase(audioFlinger, id),
+ mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
+ mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
+ mDevice(device)
+{
+ readOutputParameters();
+
+ mMasterVolume = mAudioFlinger->masterVolume();
+ mMasterMute = mAudioFlinger->masterMute();
+
+ for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+ mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
+ mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
+ }
+}
+
+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 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 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::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;
+}
+
+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()
+{
+ if (mSampleRate == 0) {
+ LOGE("No working audio driver found.");
+ return NO_INIT;
+ }
+ LOGI("AudioFlinger's thread %p ready to run", this);
+ return NO_ERROR;
+}
+
+void AudioFlinger::PlaybackThread::onFirstRef()
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+
+ snprintf(buffer, SIZE, "Playback Thread %p", this);
+
+ run(buffer, 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,
+ int format,
+ int channelCount,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId,
+ status_t *status)
+{
+ sp<Track> track;
+ status_t lStatus;
+
+ if (mType == DIRECT) {
+ if (sampleRate != mSampleRate || format != mFormat || channelCount != (int)mChannelCount) {
+ LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p",
+ sampleRate, format, channelCount, mOutput);
+ 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;
+ }
+ }
+
+ if (mOutput == 0) {
+ LOGE("Audio driver not initialized.");
+ lStatus = NO_INIT;
+ 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((AudioSystem::stream_type)streamType);
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ sp<Track> t = mTracks[i];
+ if (t != 0) {
+ if (sessionId == t->sessionId() &&
+ strategy != AudioSystem::getStrategyForStream((AudioSystem::stream_type)t->type())) {
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+ }
+ }
+
+ track = new Track(this, client, streamType, sampleRate, format,
+ channelCount, 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) {
+ LOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
+ track->setMainBuffer(chain->inBuffer());
+ chain->setStrategy(AudioSystem::getStrategyForStream((AudioSystem::stream_type)track->type()));
+ }
+ }
+ lStatus = NO_ERROR;
+
+Exit:
+ if(status) {
+ *status = lStatus;
+ }
+ return track;
+}
+
+uint32_t AudioFlinger::PlaybackThread::latency() const
+{
+ if (mOutput) {
+ return mOutput->latency();
+ }
+ else {
+ return 0;
+ }
+}
+
+status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
+{
+#ifdef LVMX
+ int audioOutputType = LifeVibes::getMixerType(mId, mType);
+ if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
+ LifeVibes::setMasterVolume(audioOutputType, value);
+ }
+#endif
+ mMasterVolume = value;
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
+{
+#ifdef LVMX
+ int audioOutputType = LifeVibes::getMixerType(mId, mType);
+ if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
+ LifeVibes::setMasterMute(audioOutputType, muted);
+ }
+#endif
+ 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)
+{
+#ifdef LVMX
+ int audioOutputType = LifeVibes::getMixerType(mId, mType);
+ if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
+ LifeVibes::setStreamVolume(audioOutputType, stream, value);
+ }
+#endif
+ mStreamTypes[stream].volume = value;
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
+{
+#ifdef LVMX
+ int audioOutputType = LifeVibes::getMixerType(mId, mType);
+ if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
+ LifeVibes::setStreamMute(audioOutputType, stream, muted);
+ }
+#endif
+ 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;
+}
+
+bool AudioFlinger::PlaybackThread::isStreamActive(int stream) const
+{
+ Mutex::Autolock _l(mLock);
+ size_t count = mActiveTracks.size();
+ for (size_t i = 0 ; i < count ; ++i) {
+ sp<Track> t = mActiveTracks[i].promote();
+ if (t == 0) continue;
+ Track* const track = t.get();
+ if (t->type() == stream)
+ return true;
+ }
+ return false;
+}
+
+// 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) {
+ LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
+ chain->startTrack();
+ }
+ }
+
+ status = NO_ERROR;
+ }
+
+ LOGV("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) {
+ mTracks.remove(track);
+ deleteTrackName_l(track->name());
+ }
+}
+
+String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
+{
+ return mOutput->getParameters(keys);
+}
+
+// destroyTrack_l() must be called with AudioFlinger::mLock held
+void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
+ AudioSystem::OutputDescriptor desc;
+ void *param2 = 0;
+
+ LOGV("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 = mChannels;
+ desc.samplingRate = mSampleRate;
+ desc.format = mFormat;
+ desc.frameCount = mFrameCount;
+ desc.latency = latency();
+ param2 = &desc;
+ break;
+
+ case AudioSystem::STREAM_CONFIG_CHANGED:
+ param2 = &param;
+ case AudioSystem::OUTPUT_CLOSED:
+ default:
+ break;
+ }
+ mAudioFlinger->audioConfigChanged_l(event, mId, param2);
+}
+
+void AudioFlinger::PlaybackThread::readOutputParameters()
+{
+ mSampleRate = mOutput->sampleRate();
+ mChannels = mOutput->channels();
+ mChannelCount = (uint16_t)AudioSystem::popCount(mChannels);
+ mFormat = mOutput->format();
+ mFrameSize = (uint16_t)mOutput->frameSize();
+ mFrameCount = mOutput->bufferSize() / 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;
+ }
+ if (mOutput == 0) {
+ return INVALID_OPERATION;
+ }
+ *halFrames = mBytesWritten/mOutput->frameSize();
+
+ return mOutput->getRenderPosition(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 AudioSystem::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 == AudioSystem::SESSION_OUTPUT_MIX) {
+ return AudioSystem::getStrategyForStream(AudioSystem::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((AudioSystem::stream_type) track->type());
+ }
+ }
+ return AudioSystem::getStrategyForStream(AudioSystem::MUSIC);
+}
+
+sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain(int sessionId)
+{
+ Mutex::Autolock _l(mLock);
+ return getEffectChain_l(sessionId);
+}
+
+sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::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::PlaybackThread::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);
+ }
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
+ : PlaybackThread(audioFlinger, output, id, device),
+ mAudioMixer(0)
+{
+ mType = PlaybackThread::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
+ nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3;
+ nsecs_t lastWarning = 0;
+ bool longStandbyExit = false;
+ uint32_t activeSleepTime = activeSleepTimeUs();
+ uint32_t idleSleepTime = idleSleepTimeUs();
+ uint32_t sleepTime = idleSleepTime;
+ Vector< sp<EffectChain> > effectChains;
+
+ while (!exitPending())
+ {
+ 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
+ maxPeriod = seconds(mFrameCount) / mSampleRate * 3;
+ activeSleepTime = activeSleepTimeUs();
+ idleSleepTime = idleSleepTimeUs();
+ }
+
+ const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
+
+ // put audio hardware into standby after short delay
+ if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
+ mSuspended) {
+ if (!mStandby) {
+ LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
+ mOutput->standby();
+ 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;
+
+ // wait until we have something to do...
+ LOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
+ mWaitWorkCV.wait(mLock);
+ LOGV("MixerThread %p TID %d waking up\n", this, gettid());
+
+ 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 (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
+ // mix buffers...
+ mAudioMixer->process();
+ sleepTime = 0;
+ 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;
+ } else {
+ sleepTime = idleSleepTime;
+ }
+ } else if (mBytesWritten != 0 ||
+ (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
+ memset (mMixBuffer, 0, mixBufferSize);
+ sleepTime = 0;
+ LOGV_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);
+#ifdef LVMX
+ int audioOutputType = LifeVibes::getMixerType(mId, mType);
+ if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
+ LifeVibes::process(audioOutputType, mMixBuffer, mixBufferSize);
+ }
+#endif
+ mLastWriteTime = systemTime();
+ mInWrite = true;
+ mBytesWritten += mixBufferSize;
+
+ int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize);
+ if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
+ mNumWrites++;
+ mInWrite = false;
+ nsecs_t now = systemTime();
+ nsecs_t delta = now - mLastWriteTime;
+ if (delta > maxPeriod) {
+ mNumDelayedWrites++;
+ if ((now - lastWarning) > kWarningThrottle) {
+ 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->standby();
+ }
+
+ LOGV("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;
+ }
+#ifdef LVMX
+ bool tracksConnectedChanged = false;
+ bool stateChanged = false;
+
+ int audioOutputType = LifeVibes::getMixerType(mId, mType);
+ if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType))
+ {
+ int activeTypes = 0;
+ for (size_t i=0 ; i<count ; i++) {
+ sp<Track> t = activeTracks[i].promote();
+ if (t == 0) continue;
+ Track* const track = t.get();
+ int iTracktype=track->type();
+ activeTypes |= 1<<track->type();
+ }
+ LifeVibes::computeVolumes(audioOutputType, activeTypes, tracksConnectedChanged, stateChanged, masterVolume, masterMute);
+ }
+#endif
+ // Delegate master volume control to effect in output mix effect chain if needed
+ sp<EffectChain> chain = getEffectChain_l(AudioSystem::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());
+ if (cblk->framesReady() && (track->isReady() || track->isStopped()) &&
+ !track->isPaused() && !track->isTerminated())
+ {
+ //LOGV("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;
+ }
+ } 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;
+#ifdef LVMX
+ bool streamMute=false;
+ // read the volume from the LivesVibes audio engine.
+ if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType))
+ {
+ LifeVibes::getStreamVolumes(audioOutputType, track->type(), &typeVolume, &streamMute);
+ if (streamMute) {
+ typeVolume = 0;
+ }
+ }
+#endif
+ 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);
+
+#ifdef LVMX
+ if ( tracksConnectedChanged || stateChanged )
+ {
+ // only do the ramp when the volume is changed by the user / application
+ param = AudioMixer::VOLUME;
+ }
+#endif
+
+ // XXX: these things DON'T need to be done each time
+ mAudioMixer->setBufferProvider(track);
+ mAudioMixer->enable(AudioMixer::MIXING);
+
+ 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_COUNT, (void *)track->channelCount());
+ 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 {
+ //LOGV("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) {
+ LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this);
+ tracksToRemove->add(track);
+ } else if (mixerStatus != MIXER_TRACKS_READY) {
+ mixerStatus = MIXER_TRACKS_ENABLED;
+ }
+ }
+ mAudioMixer->disable(AudioMixer::MIXING);
+ }
+ }
+
+ // remove all the tracks that need to be...
+ count = tracksToRemove->size();
+ if (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) {
+ LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
+ chain->stopTrack();
+ }
+ }
+ if (track->isTerminated()) {
+ mTracks.remove(track);
+ deleteTrackName_l(track->mName);
+ }
+ }
+ }
+
+ // 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)
+{
+ LOGV ("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) {
+ t->mCblk->lock.lock();
+ t->mCblk->flags |= CBLK_INVALID_ON;
+ t->mCblk->cv.signal();
+ t->mCblk->lock.unlock();
+ }
+ }
+}
+
+
+// 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)
+{
+ LOGV("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 != AudioSystem::PCM_16_BIT) {
+ status = BAD_VALUE;
+ } else {
+ reconfig = true;
+ }
+ }
+ if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
+ if (value != AudioSystem::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 garantied
+ // 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) {
+ // 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->setParameters(keyValuePair);
+ if (!mStandby && status == INVALID_OPERATION) {
+ mOutput->standby();
+ mStandby = true;
+ mBytesWritten = 0;
+ status = mOutput->setParameters(keyValuePair);
+ }
+ 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();
+ mWaitWorkCV.wait(mLock);
+ }
+ 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::activeSleepTimeUs()
+{
+ return (uint32_t)(mOutput->latency() * 1000) / 2;
+}
+
+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 = PlaybackThread::DIRECT;
+}
+
+AudioFlinger::DirectOutputThread::~DirectOutputThread()
+{
+}
+
+
+static inline int16_t clamp16(int32_t sample)
+{
+ if ((sample>>15) ^ (sample>>31))
+ sample = 0x7FFF ^ (sample>>31);
+ return sample;
+}
+
+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 (!AudioSystem::isLinearPCM(mFormat)) {
+ return;
+ }
+
+ // convert to signed 16 bit before volume calculation
+ if (mFormat == AudioSystem::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 == AudioSystem::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);
+
+ 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 UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
+ mSuspended) {
+ // wait until we have something to do...
+ if (!mStandby) {
+ LOGV("Audio hardware entering standby, mixer %p\n", this);
+ mOutput->standby();
+ 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;
+
+ LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
+ mWaitWorkCV.wait(mLock);
+ LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
+
+ 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->isStopped()) &&
+ !track->isPaused() && !track->isTerminated())
+ {
+ //LOGV("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->setVolume(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 {
+ //LOGV("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) {
+ LOGV("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 (UNLIKELY(trackToRemove != 0)) {
+ mActiveTracks.remove(trackToRemove);
+ if (!effectChains.isEmpty()) {
+ LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
+ trackToRemove->sessionId());
+ effectChains[0]->stopTrack();
+ }
+ if (trackToRemove->isTerminated()) {
+ mTracks.remove(trackToRemove);
+ deleteTrackName_l(trackToRemove->mName);
+ }
+ }
+
+ lockEffectChains_l(effectChains);
+ }
+
+ if (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 (UNLIKELY(buffer.raw == 0)) {
+ 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 && AudioSystem::isLinearPCM(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->write(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->standby();
+ }
+
+ LOGV("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->setParameters(keyValuePair);
+ if (!mStandby && status == INVALID_OPERATION) {
+ mOutput->standby();
+ mStandby = true;
+ mBytesWritten = 0;
+ status = mOutput->setParameters(keyValuePair);
+ }
+ if (status == NO_ERROR && reconfig) {
+ readOutputParameters();
+ sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
+ }
+ }
+
+ mNewParameters.removeAt(0);
+
+ mParamStatus = status;
+ mParamCond.signal();
+ mWaitWorkCV.wait(mLock);
+ }
+ return reconfig;
+}
+
+uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
+{
+ uint32_t time;
+ if (AudioSystem::isLinearPCM(mFormat)) {
+ time = (uint32_t)(mOutput->latency() * 1000) / 2;
+ } else {
+ time = 10000;
+ }
+ return time;
+}
+
+uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
+{
+ uint32_t time;
+ if (AudioSystem::isLinearPCM(mFormat)) {
+ time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
+ } else {
+ time = 10000;
+ }
+ return time;
+}
+
+uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
+{
+ uint32_t time;
+ if (AudioSystem::isLinearPCM(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 = PlaybackThread::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;
+
+ 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 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;
+
+ LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
+ mWaitWorkCV.wait(mLock);
+ LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
+ 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 (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();
+ }
+
+ return false;
+}
+
+void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
+{
+ int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
+ OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
+ this,
+ mSampleRate,
+ mFormat,
+ mChannelCount,
+ frameCount);
+ if (outputTrack->cblk() != NULL) {
+ thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f);
+ mOutputTracks.add(outputTrack);
+ LOGV("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;
+ }
+ }
+ LOGV("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()) {
+ LOGV("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,
+ int format,
+ int channelCount,
+ 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)
+{
+ LOGV_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);
+ 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;
+ mCblk->channelCount = (uint8_t)channelCount;
+ 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
+ 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;
+ mCblk->channelCount = (uint8_t)channelCount;
+ 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
+ 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 = 0;
+ 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) {
+ LOGV("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);
+ LOGV("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 (int)mCblk->channelCount;
+}
+
+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, channelCount %d",
+ bufferStart, bufferEnd, mBuffer, mBufferEnd,
+ cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount);
+ 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,
+ int format,
+ int channelCount,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId)
+ : TrackBase(thread, client, sampleRate, format, channelCount, 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();
+ }
+ LOGV("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 = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t);
+ }
+}
+
+AudioFlinger::PlaybackThread::Track::~Track()
+{
+ LOGV("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(),
+ (AudioSystem::stream_type)mStreamType,
+ mSessionId);
+ }
+ 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 %03u %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,
+ mCblk->channelCount,
+ 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;
+ LOGV("stepServer recovered");
+ mFlags &= ~TrackBase::STEPSERVER_FAILED;
+ }
+
+ framesReady = cblk->framesReady();
+
+ if (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 == 0) goto getNextBuffer_exit;
+
+ buffer->frameCount = framesReq;
+ return NO_ERROR;
+ }
+
+getNextBuffer_exit:
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ LOGV("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) return true;
+
+ if (mCblk->framesReady() >= mCblk->frameCount ||
+ (mCblk->flags & CBLK_FORCEREADY_MSK)) {
+ mFillingUpStatus = FS_FILLED;
+ mCblk->flags &= ~CBLK_FORCEREADY_MSK;
+ return true;
+ }
+ return false;
+}
+
+status_t AudioFlinger::PlaybackThread::Track::start()
+{
+ status_t status = NO_ERROR;
+ LOGV("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;
+ LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
+ } else {
+ mState = TrackBase::ACTIVE;
+ LOGV("? => ACTIVE (%d) on thread %p", mName, this);
+ }
+
+ if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
+ thread->mLock.unlock();
+ status = AudioSystem::startOutput(thread->id(),
+ (AudioSystem::stream_type)mStreamType,
+ mSessionId);
+ thread->mLock.lock();
+ }
+ 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()
+{
+ LOGV("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();
+ }
+ LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
+ }
+ if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
+ thread->mLock.unlock();
+ AudioSystem::stopOutput(thread->id(),
+ (AudioSystem::stream_type)mStreamType,
+ mSessionId);
+ thread->mLock.lock();
+ }
+ }
+}
+
+void AudioFlinger::PlaybackThread::Track::pause()
+{
+ LOGV("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;
+ LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
+ if (!isOutputTrack()) {
+ thread->mLock.unlock();
+ AudioSystem::stopOutput(thread->id(),
+ (AudioSystem::stream_type)mStreamType,
+ mSessionId);
+ thread->mLock.lock();
+ }
+ }
+ }
+}
+
+void AudioFlinger::PlaybackThread::Track::flush()
+{
+ LOGV("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;
+
+ mCblk->lock.lock();
+ // NOTE: reset() will reset cblk->user and cblk->server with
+ // the risk that at the same time, the AudioMixer is trying to read
+ // data. In this case, getNextBuffer() would return a NULL pointer
+ // as audio buffer => the AudioMixer code MUST always test that pointer
+ // returned by getNextBuffer() is not NULL!
+ reset();
+ mCblk->lock.unlock();
+ }
+}
+
+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
+ mCblk->flags |= CBLK_UNDERRUN_ON;
+ mCblk->flags &= ~CBLK_FORCEREADY_MSK;
+ 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,
+ int format,
+ int channelCount,
+ int frameCount,
+ uint32_t flags,
+ int sessionId)
+ : TrackBase(thread, client, sampleRate, format,
+ channelCount, frameCount, flags, 0, sessionId),
+ mOverflow(false)
+{
+ if (mCblk != NULL) {
+ LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
+ if (format == AudioSystem::PCM_16_BIT) {
+ mCblk->frameSize = channelCount * sizeof(int16_t);
+ } else if (format == AudioSystem::PCM_8_BIT) {
+ mCblk->frameSize = channelCount * 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;
+ LOGV("stepServer recovered");
+ mFlags &= ~TrackBase::STEPSERVER_FAILED;
+ }
+
+ framesAvail = cblk->framesAvailable_l();
+
+ if (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 == 0) goto getNextBuffer_exit;
+
+ buffer->frameCount = framesReq;
+ return NO_ERROR;
+ }
+
+getNextBuffer_exit:
+ buffer->raw = 0;
+ 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
+ mCblk->flags |= CBLK_UNDERRUN_ON;
+ }
+}
+
+void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
+{
+ snprintf(buffer, size, " %05d %03u %03u %05d %04u %01d %05u %08x %08x\n",
+ (mClient == NULL) ? getpid() : mClient->pid(),
+ mFormat,
+ mCblk->channelCount,
+ mSessionId,
+ mFrameCount,
+ mState,
+ mCblk->sampleRate,
+ mCblk->server,
+ mCblk->user);
+}
+
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
+ const wp<ThreadBase>& thread,
+ DuplicatingThread *sourceThread,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount)
+ : Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, 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);
+ LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p",
+ mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, 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 = mCblk->channelCount;
+ 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) {
+ LOGV ("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;
+ LOGV("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);
+ LOGV("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;
+
+// LOGV("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 (UNLIKELY(!active)) {
+ LOGV("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,
+ int format,
+ int channelCount,
+ 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 != AudioSystem::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 = new RecordThread::RecordTrack(thread, client, sampleRate,
+ format, channelCount, frameCount, flags, lSessionId);
+ }
+ if (recordTrack->getCblk() == NULL) {
+ // 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();
+ lStatus = NO_MEMORY;
+ 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() {
+ LOGV("RecordHandle::start()");
+ return mRecordTrack->start();
+}
+
+void AudioFlinger::RecordHandle::stop() {
+ LOGV("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) :
+ ThreadBase(audioFlinger, id),
+ mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
+{
+ mReqChannelCount = AudioSystem::popCount(channels);
+ mReqSampleRate = sampleRate;
+ readInputParameters();
+}
+
+
+AudioFlinger::RecordThread::~RecordThread()
+{
+ delete[] mRsmpInBuffer;
+ if (mResampler != 0) {
+ delete mResampler;
+ delete[] mRsmpOutBuffer;
+ }
+}
+
+void AudioFlinger::RecordThread::onFirstRef()
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+
+ snprintf(buffer, SIZE, "Record Thread %p", this);
+
+ run(buffer, PRIORITY_URGENT_AUDIO);
+}
+
+bool AudioFlinger::RecordThread::threadLoop()
+{
+ AudioBufferProvider::Buffer buffer;
+ sp<RecordTrack> activeTrack;
+
+ // start recording
+ while (!exitPending()) {
+
+ processConfigEvents();
+
+ { // scope for mLock
+ Mutex::Autolock _l(mLock);
+ checkForNewParameters_l();
+ if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
+ if (!mStandby) {
+ mInput->standby();
+ mStandby = true;
+ }
+
+ if (exitPending()) break;
+
+ LOGV("RecordThread: loop stopping");
+ // go to sleep
+ mWaitWorkCV.wait(mLock);
+ LOGV("RecordThread: loop starting");
+ continue;
+ }
+ if (mActiveTrack != 0) {
+ if (mActiveTrack->mState == TrackBase::PAUSING) {
+ if (!mStandby) {
+ mInput->standby();
+ 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;
+ }
+ }
+ }
+
+ if (mActiveTrack != 0) {
+ if (mActiveTrack->mState != TrackBase::ACTIVE &&
+ mActiveTrack->mState != TrackBase::RESUMING) {
+ usleep(5000);
+ continue;
+ }
+ buffer.frameCount = mFrameCount;
+ if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+ size_t framesOut = buffer.frameCount;
+ if (mResampler == 0) {
+ // 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 != AudioSystem::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 != AudioSystem::PCM_16_BIT)) {
+ mBytesRead = mInput->read(buffer.raw, mInputBytes);
+ framesOut = 0;
+ } else {
+ mBytesRead = mInput->read(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->standby();
+ usleep(5000);
+ }
+ 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) {
+ AudioMixer::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 {
+ AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
+ }
+
+ }
+ mActiveTrack->releaseBuffer(&buffer);
+ mActiveTrack->overflow();
+ }
+ // client isn't retrieving buffers fast enough
+ else {
+ if (!mActiveTrack->setOverflow())
+ LOGW("RecordThread: buffer overflow");
+ // 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(5000);
+ }
+ }
+ }
+
+ if (!mStandby) {
+ mInput->standby();
+ }
+ mActiveTrack.clear();
+
+ mStartStopCond.broadcast();
+
+ LOGV("RecordThread %p exiting", this);
+ return false;
+}
+
+status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
+{
+ LOGV("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;
+ }
+ mActiveTrack->mState = TrackBase::RESUMING;
+ mRsmpInIndex = mFrameCount;
+ mBytesRead = 0;
+ // signal thread to start
+ LOGV("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) {
+ LOGV("Record failed to start");
+ status = BAD_VALUE;
+ goto startError;
+ }
+ LOGV("Record started OK");
+ return status;
+ }
+startError:
+ AudioSystem::stopInput(mId);
+ return status;
+}
+
+void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
+ LOGV("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();
+ LOGV("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 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 != 0));
+ 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);
+
+ 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->read(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->standby();
+ usleep(5000);
+ }
+ buffer->raw = 0;
+ 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 = AudioSystem::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 (status == NO_ERROR) {
+ status = mInput->setParameters(keyValuePair);
+ if (status == INVALID_OPERATION) {
+ mInput->standby();
+ status = mInput->setParameters(keyValuePair);
+ }
+ if (reconfig) {
+ if (status == BAD_VALUE &&
+ reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT &&
+ ((int)mInput->sampleRate() <= 2 * reqSamplingRate) &&
+ (AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) {
+ status = NO_ERROR;
+ }
+ if (status == NO_ERROR) {
+ readInputParameters();
+ sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
+ }
+ }
+ }
+
+ mNewParameters.removeAt(0);
+
+ mParamStatus = status;
+ mParamCond.signal();
+ mWaitWorkCV.wait(mLock);
+ }
+ return reconfig;
+}
+
+String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
+{
+ return mInput->getParameters(keys);
+}
+
+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 = mChannels;
+ 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 = 0;
+
+ mSampleRate = mInput->sampleRate();
+ mChannels = mInput->channels();
+ mChannelCount = (uint16_t)AudioSystem::popCount(mChannels);
+ mFormat = mInput->format();
+ mFrameSize = (uint16_t)mInput->frameSize();
+ mInputBytes = mInput->bufferSize();
+ 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()
+{
+ return mInput->getInputFramesLost();
+}
+
+// ----------------------------------------------------------------------------
+
+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;
+
+ LOGV("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);
+
+ AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices,
+ (int *)&format,
+ &channels,
+ &samplingRate,
+ &status);
+ LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
+ output,
+ samplingRate,
+ format,
+ channels,
+ status);
+
+ mHardwareStatus = AUDIO_HW_IDLE;
+ if (output != 0) {
+ int id = nextUniqueId();
+ if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
+ (format != AudioSystem::PCM_16_BIT) ||
+ (channels != AudioSystem::CHANNEL_OUT_STEREO)) {
+ thread = new DirectOutputThread(this, output, id, *pDevices);
+ LOGV("openOutput() created direct output: ID %d thread %p", id, thread);
+ } else {
+ thread = new MixerThread(this, output, id, *pDevices);
+ LOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
+
+#ifdef LVMX
+ unsigned bitsPerSample =
+ (format == AudioSystem::PCM_16_BIT) ? 16 :
+ ((format == AudioSystem::PCM_8_BIT) ? 8 : 0);
+ unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1;
+ int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id());
+
+ LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount);
+ LifeVibes::setDevice(audioOutputType, *pDevices);
+#endif
+
+ }
+ 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;
+ }
+
+ LOGV("closeOutput() %d", output);
+
+ if (thread->type() == PlaybackThread::MIXER) {
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::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() != PlaybackThread::DUPLICATING) {
+ mAudioHardware->closeOutputStream(thread->getOutput());
+ }
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::suspendOutput(int output)
+{
+ Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+
+ if (thread == NULL) {
+ return BAD_VALUE;
+ }
+
+ LOGV("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;
+ }
+
+ LOGV("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;
+
+ if (pDevices == NULL || *pDevices == 0) {
+ return 0;
+ }
+ Mutex::Autolock _l(mLock);
+
+ AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices,
+ (int *)&format,
+ &channels,
+ &samplingRate,
+ &status,
+ (AudioSystem::audio_in_acoustics)acoustics);
+ LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
+ input,
+ 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 (input == 0 && status == BAD_VALUE &&
+ reqFormat == format && format == AudioSystem::PCM_16_BIT &&
+ (samplingRate <= 2 * reqSamplingRate) &&
+ (AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) {
+ LOGV("openInput() reopening with proposed sampling rate and channels");
+ input = mAudioHardware->openInputStream(*pDevices,
+ (int *)&format,
+ &channels,
+ &samplingRate,
+ &status,
+ (AudioSystem::audio_in_acoustics)acoustics);
+ }
+
+ if (input != 0) {
+ int id = nextUniqueId();
+ // Start record thread
+ thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id);
+ mRecordThreads.add(id, thread);
+ LOGV("openInput() created record thread: ID %d thread %p", id, thread);
+ if (pSamplingRate) *pSamplingRate = reqSamplingRate;
+ if (pFormat) *pFormat = format;
+ if (pChannels) *pChannels = reqChannels;
+
+ input->standby();
+
+ // 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;
+ }
+
+ LOGV("closeInput() %d", input);
+ void *param2 = 0;
+ audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
+ mRecordThreads.removeItem(input);
+ }
+ thread->exit();
+
+ mAudioHardware->closeInputStream(thread->getInput());
+
+ 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;
+ }
+
+ LOGV("setStreamOutput() stream %d to output %d", stream, output);
+ audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
+
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
+ if (thread != dstThread &&
+ thread->type() != PlaybackThread::DIRECT) {
+ MixerThread *srcThread = (MixerThread *)thread;
+ srcThread->invalidateTracks(stream);
+ }
+ }
+
+ return NO_ERROR;
+}
+
+
+int AudioFlinger::newAudioSessionId()
+{
+ return nextUniqueId();
+}
+
+// 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() == PlaybackThread::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;
+}
+
+int AudioFlinger::nextUniqueId()
+{
+ return android_atomic_inc(&mNextUniqueId);
+}
+
+// ----------------------------------------------------------------------------
+// Effect management
+// ----------------------------------------------------------------------------
+
+
+status_t AudioFlinger::loadEffectLibrary(const char *libPath, int *handle)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+ // only allow libraries loaded from /system/lib/soundfx for now
+ if (strncmp(gEffectLibPath, libPath, strlen(gEffectLibPath)) != 0) {
+ return PERMISSION_DENIED;
+ }
+
+ Mutex::Autolock _l(mLock);
+ return EffectLoadLibrary(libPath, handle);
+}
+
+status_t AudioFlinger::unloadEffectLibrary(int handle)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+
+ Mutex::Autolock _l(mLock);
+ return EffectUnloadLibrary(handle);
+}
+
+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);
+}
+
+
+// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp
+static const effect_uuid_t VISUALIZATION_UUID_ =
+ {0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}};
+
+sp<IEffect> AudioFlinger::createEffect(pid_t pid,
+ effect_descriptor_t *pDesc,
+ const sp<IEffectClient>& effectClient,
+ int32_t priority,
+ int output,
+ int sessionId,
+ status_t *status,
+ int *id,
+ int *enabled)
+{
+ status_t lStatus = NO_ERROR;
+ sp<EffectHandle> handle;
+ effect_interface_t itfe;
+ effect_descriptor_t desc;
+ sp<Client> client;
+ wp<Client> wclient;
+
+ LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, output %d",
+ pid, effectClient.get(), priority, sessionId, output);
+
+ if (pDesc == NULL) {
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+
+ {
+ Mutex::Autolock _l(mLock);
+
+ // check recording permission for visualizer
+ if (memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 ||
+ memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) {
+ if (!recordingAllowed()) {
+ lStatus = PERMISSION_DENIED;
+ goto Exit;
+ }
+ }
+
+ 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;
+ 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 != AudioSystem::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 == AudioSystem::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 != AudioSystem::SESSION_OUTPUT_MIX &&
+ (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+ lStatus = INVALID_OPERATION;
+ goto Exit;
+ }
+
+ // Session AudioSystem::SESSION_OUTPUT_STAGE is reserved for output stage effects
+ // that can only be created by audio policy manager (running in same process)
+ if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE &&
+ getpid() != pid) {
+ lStatus = INVALID_OPERATION;
+ 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.
+ // TODO: allow attachment of effect to inputs
+ if (output == 0) {
+ if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE) {
+ // output must be specified by AudioPolicyManager when using session
+ // AudioSystem::SESSION_OUTPUT_STAGE
+ lStatus = BAD_VALUE;
+ goto Exit;
+ } else if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) {
+ output = AudioSystem::getOutputForEffect(&desc);
+ LOGV("createEffect() got output %d for effect %s", output, desc.name);
+ } else {
+ // 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) {
+ output = mPlaybackThreads.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 (output == 0 && mPlaybackThreads.size()) {
+ output = mPlaybackThreads.keyAt(0);
+ }
+ }
+ }
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ 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 trhead
+ 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 session, int srcOutput, int dstOutput)
+{
+ LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
+ session, 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(session, srcThread, dstThread, false);
+
+ return NO_ERROR;
+}
+
+// moveEffectChain_l mustbe called with both srcThread and dstThread mLocks held
+status_t AudioFlinger::moveEffectChain_l(int session,
+ AudioFlinger::PlaybackThread *srcThread,
+ AudioFlinger::PlaybackThread *dstThread,
+ bool reRegister)
+{
+ LOGV("moveEffectChain_l() session %d from thread %p to thread %p",
+ session, srcThread, dstThread);
+
+ sp<EffectChain> chain = srcThread->getEffectChain_l(session);
+ if (chain == 0) {
+ LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
+ session, 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 unecessary 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;
+ sp<EffectModule> effect = chain->getEffectFromId_l(0);
+ while (effect != 0) {
+ srcThread->removeEffect_l(effect);
+ dstThread->addEffect_l(effect);
+ // 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,
+ session,
+ effect->id());
+ }
+ effect = chain->getEffectFromId_l(0);
+ }
+
+ return NO_ERROR;
+}
+
+// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
+sp<AudioFlinger::EffectHandle> AudioFlinger::PlaybackThread::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<Track> track;
+ sp<EffectChain> chain;
+ bool chainCreated = false;
+ bool effectCreated = false;
+ bool effectRegistered = false;
+
+ if (mOutput == 0) {
+ LOGW("createEffect_l() Audio driver not initialized.");
+ lStatus = NO_INIT;
+ goto Exit;
+ }
+
+ // Do not allow auxiliary effect on session other than 0
+ if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY &&
+ sessionId != AudioSystem::SESSION_OUTPUT_MIX) {
+ LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
+ desc->name, sessionId);
+ lStatus = BAD_VALUE;
+ 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 == AudioSystem::SESSION_OUTPUT_MIX && mType != MIXER) {
+ LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
+ desc->name, sessionId);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+
+ LOGV("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
+ LOGV("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);
+ }
+
+ LOGV("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;
+}
+
+// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
+// PlaybackThread::mLock held
+status_t AudioFlinger::PlaybackThread::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
+ LOGV("addEffect_l() new effect chain for session %d", sessionId);
+ chain = new EffectChain(this, sessionId);
+ addEffectChain_l(chain);
+ chain->setStrategy(getStrategyForSession_l(sessionId));
+ chainCreated = true;
+ }
+ LOGV("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::PlaybackThread::removeEffect_l(const sp<EffectModule>& effect) {
+
+ LOGV("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::PlaybackThread::disconnectEffect(const sp<EffectModule>& effect,
+ const wp<EffectHandle>& handle) {
+ Mutex::Autolock _l(mLock);
+ LOGV("disconnectEffect() %p effect %p", this, effect.get());
+ // delete the effect module if removing last handle on it
+ if (effect->removeHandle(handle) == 0) {
+ 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;
+
+ LOGV("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));
+ LOGV("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()) {
+ LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
+ track->setMainBuffer(buffer);
+ }
+ }
+
+ // 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()) {
+ LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
+ chain->startTrack();
+ }
+ }
+ }
+
+ chain->setInBuffer(buffer, ownsBuffer);
+ chain->setOutBuffer(mMixBuffer);
+ // Effect chain for session AudioSystem::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 AudioSystem::SESSION_OUTPUT_MIX is inserted before
+ // session AudioSystem::SESSION_OUTPUT_STAGE to be processed
+ // after track specific effects and before output stage
+ // It is therefore mandatory that AudioSystem::SESSION_OUTPUT_MIX == 0 and
+ // that AudioSystem::SESSION_OUTPUT_STAGE < AudioSystem::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);
+
+ return NO_ERROR;
+}
+
+size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
+{
+ int session = chain->sessionId();
+
+ LOGV("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 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);
+ }
+ }
+ break;
+ }
+ }
+ return mEffectChains.size();
+}
+
+void AudioFlinger::PlaybackThread::lockEffectChains_l(
+ Vector<sp <AudioFlinger::EffectChain> >& effectChains)
+{
+ effectChains = mEffectChains;
+ for (size_t i = 0; i < mEffectChains.size(); i++) {
+ mEffectChains[i]->lock();
+ }
+}
+
+void AudioFlinger::PlaybackThread::unlockEffectChains(
+ Vector<sp <AudioFlinger::EffectChain> >& effectChains)
+{
+ for (size_t i = 0; i < effectChains.size(); i++) {
+ effectChains[i]->unlock();
+ }
+}
+
+
+sp<AudioFlinger::EffectModule> AudioFlinger::PlaybackThread::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;
+}
+
+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 AudioSystem::SESSION_OUTPUT_MIX
+ sp<EffectModule> effect = getEffect_l(AudioSystem::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);
+ }
+ }
+}
+
+// ----------------------------------------------------------------------------
+// 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)
+{
+ LOGV("Constructor %p", this);
+ int lStatus;
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread == 0) {
+ return;
+ }
+ PlaybackThread *p = (PlaybackThread *)thread.get();
+
+ memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
+
+ // create effect engine from effect factory
+ mStatus = EffectCreate(&desc->uuid, sessionId, p->id(), &mEffectInterface);
+
+ if (mStatus != NO_ERROR) {
+ return;
+ }
+ lStatus = init();
+ if (lStatus < 0) {
+ mStatus = lStatus;
+ goto Error;
+ }
+
+ LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
+ return;
+Error:
+ EffectRelease(mEffectInterface);
+ mEffectInterface = NULL;
+ LOGV("Constructor Error %d", mStatus);
+}
+
+AudioFlinger::EffectModule::~EffectModule()
+{
+ LOGV("Destructor %p", this);
+ if (mEffectInterface != NULL) {
+ // 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) {
+ if (h != 0) {
+ h->setControl(false, true);
+ }
+ handle->setControl(true, false);
+ status = NO_ERROR;
+ } else {
+ status = ALREADY_EXISTS;
+ }
+ 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;
+ }
+ 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, true);
+ }
+ }
+
+ return size;
+}
+
+void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle)
+{
+ // 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) {
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ playbackThread->disconnectEffect(keep, handle);
+ }
+ }
+}
+
+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
+ break;
+ }
+}
+
+void AudioFlinger::EffectModule::process()
+{
+ Mutex::Autolock _l(mLock);
+
+ if (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) {
+ AudioMixer::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, copy input to
+ // output
+ sp<EffectChain> chain = mChain.promote();
+ if (chain != 0 && chain->activeTracks() != 0) {
+ size_t size = mConfig.inputCfg.buffer.frameCount * sizeof(int16_t);
+ if (mConfig.inputCfg.channels == CHANNEL_STEREO) {
+ size *= 2;
+ }
+ memcpy(mConfig.outputCfg.buffer.raw, mConfig.inputCfg.buffer.raw, size);
+ }
+ }
+}
+
+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 = CHANNEL_MONO;
+ } else {
+ channels = CHANNEL_STEREO;
+ }
+
+ if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
+ mConfig.inputCfg.channels = CHANNEL_MONO;
+ } else {
+ mConfig.inputCfg.channels = channels;
+ }
+ mConfig.outputCfg.channels = channels;
+ mConfig.inputCfg.format = SAMPLE_FORMAT_PCM_S15;
+ mConfig.outputCfg.format = SAMPLE_FORMAT_PCM_S15;
+ 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 AudioSystem::SESSION_OUTPUT_MIX or AudioSystem::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;
+
+ LOGV("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_CONFIGURE,
+ 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_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;
+ }
+ return status;
+}
+
+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;
+ }
+ return status;
+}
+
+status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
+ uint32_t cmdSize,
+ void *pCmdData,
+ uint32_t *replySize,
+ void *pReplyData)
+{
+ Mutex::Autolock _l(mLock);
+// LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
+
+ if (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);
+ LOGV("setEnabled %p enabled %d", this, enabled);
+
+ if (enabled != isEnabled()) {
+ 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;
+ }
+ 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:
+ default:
+ return false;
+ }
+}
+
+bool AudioFlinger::EffectModule::isProcessEnabled()
+{
+ switch (mState) {
+ case RESTART:
+ case ACTIVE:
+ case STOPPING:
+ case STOPPED:
+ return true;
+ case IDLE:
+ case STARTING:
+ 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 ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
+ // convert device bit field from AudioSystem to EffectApi format.
+ device = deviceAudioSystemToEffectApi(device);
+ if (device == 0) {
+ return BAD_VALUE;
+ }
+ status_t cmdStatus;
+ uint32_t size = sizeof(status_t);
+ status = (*mEffectInterface)->command(mEffectInterface,
+ EFFECT_CMD_SET_DEVICE,
+ sizeof(uint32_t),
+ &device,
+ &size,
+ &cmdStatus);
+ if (status == NO_ERROR) {
+ status = cmdStatus;
+ }
+ }
+ 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) {
+ // convert audio mode from AudioSystem to EffectApi format.
+ int effectMode = modeAudioSystemToEffectApi(mode);
+ if (effectMode < 0) {
+ return BAD_VALUE;
+ }
+ status_t cmdStatus;
+ uint32_t size = sizeof(status_t);
+ status = (*mEffectInterface)->command(mEffectInterface,
+ EFFECT_CMD_SET_AUDIO_MODE,
+ sizeof(int),
+ &effectMode,
+ &size,
+ &cmdStatus);
+ if (status == NO_ERROR) {
+ status = cmdStatus;
+ }
+ }
+ return status;
+}
+
+// update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified
+const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = {
+ DEVICE_EARPIECE, // AudioSystem::DEVICE_OUT_EARPIECE
+ DEVICE_SPEAKER, // AudioSystem::DEVICE_OUT_SPEAKER
+ DEVICE_WIRED_HEADSET, // case AudioSystem::DEVICE_OUT_WIRED_HEADSET
+ DEVICE_WIRED_HEADPHONE, // AudioSystem::DEVICE_OUT_WIRED_HEADPHONE
+ DEVICE_BLUETOOTH_SCO, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO
+ DEVICE_BLUETOOTH_SCO_HEADSET, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET
+ DEVICE_BLUETOOTH_SCO_CARKIT, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT
+ DEVICE_BLUETOOTH_A2DP, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP
+ DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES
+ DEVICE_BLUETOOTH_A2DP_SPEAKER, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
+ DEVICE_AUX_DIGITAL // AudioSystem::DEVICE_OUT_AUX_DIGITAL
+};
+
+uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device)
+{
+ uint32_t deviceOut = 0;
+ while (device) {
+ const uint32_t i = 31 - __builtin_clz(device);
+ device &= ~(1 << i);
+ if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) {
+ LOGE("device convertion error for AudioSystem device 0x%08x", device);
+ return 0;
+ }
+ deviceOut |= (uint32_t)sDeviceConvTable[i];
+ }
+ return deviceOut;
+}
+
+// update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified
+const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = {
+ AUDIO_MODE_NORMAL, // AudioSystem::MODE_NORMAL
+ AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE
+ AUDIO_MODE_IN_CALL // AudioSystem::MODE_IN_CALL
+};
+
+int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode)
+{
+ int modeOut = -1;
+ if (mode < sizeof(sModeConvTable) / sizeof(uint32_t)) {
+ modeOut = (int)sModeConvTable[mode];
+ }
+ return modeOut;
+}
+
+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: %04X\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), mPriority(priority), mHasControl(false)
+{
+ LOGV("constructor %p", this);
+
+ 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()
+{
+ LOGV("Destructor %p", this);
+ disconnect();
+}
+
+status_t AudioFlinger::EffectHandle::enable()
+{
+ if (!mHasControl) return INVALID_OPERATION;
+ if (mEffect == 0) return DEAD_OBJECT;
+
+ return mEffect->setEnabled(true);
+}
+
+status_t AudioFlinger::EffectHandle::disable()
+{
+ if (!mHasControl) return INVALID_OPERATION;
+ if (mEffect == NULL) return DEAD_OBJECT;
+
+ return mEffect->setEnabled(false);
+}
+
+void AudioFlinger::EffectHandle::disconnect()
+{
+ if (mEffect == 0) {
+ return;
+ }
+ mEffect->disconnect(this);
+ // release sp on module => module destructor can be called now
+ mEffect.clear();
+ if (mCblk) {
+ mCblk->~effect_param_cblk_t(); // destroy our shared-structure.
+ }
+ mCblkMemory.clear(); // and free the shared memory
+ if (mClient != 0) {
+ 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)
+{
+// LOGV("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;
+
+ // 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)
+{
+ LOGV("setControl %p control %d", this, hasControl);
+
+ mHasControl = hasControl;
+ 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 = tryLock(mCblk->lock);
+
+ snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n",
+ (mClient == NULL) ? getpid() : mClient->pid(),
+ mPriority,
+ mHasControl,
+ !locked,
+ mCblk->clientIndex,
+ mCblk->serverIndex
+ );
+
+ 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), mOwnInBuffer(false),
+ mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
+ mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
+{
+ mStrategy = AudioSystem::getStrategyForStream(AudioSystem::MUSIC);
+}
+
+AudioFlinger::EffectChain::~EffectChain()
+{
+ if (mOwnInBuffer) {
+ delete mInBuffer;
+ }
+
+}
+
+// getEffectFromDesc_l() must be called with PlaybackThread::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 PlaybackThread::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;
+}
+
+// Must be called with EffectChain::mLock locked
+void AudioFlinger::EffectChain::process_l()
+{
+ size_t size = mEffects.size();
+ for (size_t i = 0; i < size; i++) {
+ mEffects[i]->process();
+ }
+ for (size_t i = 0; i < size; i++) {
+ mEffects[i]->updateState();
+ }
+ // if no track is active, input buffer must be cleared here as the mixer process
+ // will not do it
+ if (mSessionId > 0 && activeTracks() == 0) {
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ size_t numSamples = thread->frameCount() * thread->channelCount();
+ memset(mInBuffer, 0, numSamples * sizeof(int16_t));
+ }
+ }
+}
+
+// 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);
+
+ LOGV("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]) {
+ 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);
+ LOGV("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;
+}
+
+#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
generated by cgit v1.2.3 (git 2.25.1) at 2025-10-31 21:40:29 +0000