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authorEric Laurent <elaurent@google.com>2012-03-26 10:47:22 -0700
committerEric Laurent <elaurent@google.com>2012-03-26 10:54:41 -0700
commit2e66a7896c9a9da3a15fc6cff9be28b4174d8719 (patch)
treebc6d8e86ae45de8f4b1a36eda9d11f774bde9bea /media/libmedia/SoundPool.cpp
parentef36d496477d1e2ae99c3cd43eee5cf7e82b0883 (diff)
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reorganize SoundPool and JetPlayer code.
Reorganize SoundPool and JetPlayer code to be ready for the creation of libmedia_native. Split SoundPool between libsoundpool (JNI) and libmedia(sound pool implementation). Remove dependencies on nativehelper/jni.h from JetPlayer. Change-Id: I130c6014173b714329929dd82c5dfb70b757a610
Diffstat (limited to 'media/libmedia/SoundPool.cpp')
-rw-r--r--media/libmedia/SoundPool.cpp908
1 files changed, 908 insertions, 0 deletions
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
new file mode 100644
index 0000000..306c57d
--- /dev/null
+++ b/media/libmedia/SoundPool.cpp
@@ -0,0 +1,908 @@
+/*
+ * Copyright (C) 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_NDEBUG 0
+#define LOG_TAG "SoundPool"
+#include <utils/Log.h>
+
+//#define USE_SHARED_MEM_BUFFER
+
+// XXX needed for timing latency
+#include <utils/Timers.h>
+
+#include <media/AudioTrack.h>
+#include <media/mediaplayer.h>
+
+#include <system/audio.h>
+
+#include "SoundPool.h"
+#include "SoundPoolThread.h"
+
+namespace android
+{
+
+int kDefaultBufferCount = 4;
+uint32_t kMaxSampleRate = 48000;
+uint32_t kDefaultSampleRate = 44100;
+uint32_t kDefaultFrameCount = 1200;
+
+SoundPool::SoundPool(int maxChannels, audio_stream_type_t streamType, int srcQuality)
+{
+ ALOGV("SoundPool constructor: maxChannels=%d, streamType=%d, srcQuality=%d",
+ maxChannels, streamType, srcQuality);
+
+ // check limits
+ mMaxChannels = maxChannels;
+ if (mMaxChannels < 1) {
+ mMaxChannels = 1;
+ }
+ else if (mMaxChannels > 32) {
+ mMaxChannels = 32;
+ }
+ ALOGW_IF(maxChannels != mMaxChannels, "App requested %d channels", maxChannels);
+
+ mQuit = false;
+ mDecodeThread = 0;
+ mStreamType = streamType;
+ mSrcQuality = srcQuality;
+ mAllocated = 0;
+ mNextSampleID = 0;
+ mNextChannelID = 0;
+
+ mCallback = 0;
+ mUserData = 0;
+
+ mChannelPool = new SoundChannel[mMaxChannels];
+ for (int i = 0; i < mMaxChannels; ++i) {
+ mChannelPool[i].init(this);
+ mChannels.push_back(&mChannelPool[i]);
+ }
+
+ // start decode thread
+ startThreads();
+}
+
+SoundPool::~SoundPool()
+{
+ ALOGV("SoundPool destructor");
+ mDecodeThread->quit();
+ quit();
+
+ Mutex::Autolock lock(&mLock);
+
+ mChannels.clear();
+ if (mChannelPool)
+ delete [] mChannelPool;
+ // clean up samples
+ ALOGV("clear samples");
+ mSamples.clear();
+
+ if (mDecodeThread)
+ delete mDecodeThread;
+}
+
+void SoundPool::addToRestartList(SoundChannel* channel)
+{
+ Mutex::Autolock lock(&mRestartLock);
+ if (!mQuit) {
+ mRestart.push_back(channel);
+ mCondition.signal();
+ }
+}
+
+void SoundPool::addToStopList(SoundChannel* channel)
+{
+ Mutex::Autolock lock(&mRestartLock);
+ if (!mQuit) {
+ mStop.push_back(channel);
+ mCondition.signal();
+ }
+}
+
+int SoundPool::beginThread(void* arg)
+{
+ SoundPool* p = (SoundPool*)arg;
+ return p->run();
+}
+
+int SoundPool::run()
+{
+ mRestartLock.lock();
+ while (!mQuit) {
+ mCondition.wait(mRestartLock);
+ ALOGV("awake");
+ if (mQuit) break;
+
+ while (!mStop.empty()) {
+ SoundChannel* channel;
+ ALOGV("Getting channel from stop list");
+ List<SoundChannel* >::iterator iter = mStop.begin();
+ channel = *iter;
+ mStop.erase(iter);
+ mRestartLock.unlock();
+ if (channel != 0) {
+ Mutex::Autolock lock(&mLock);
+ channel->stop();
+ }
+ mRestartLock.lock();
+ if (mQuit) break;
+ }
+
+ while (!mRestart.empty()) {
+ SoundChannel* channel;
+ ALOGV("Getting channel from list");
+ List<SoundChannel*>::iterator iter = mRestart.begin();
+ channel = *iter;
+ mRestart.erase(iter);
+ mRestartLock.unlock();
+ if (channel != 0) {
+ Mutex::Autolock lock(&mLock);
+ channel->nextEvent();
+ }
+ mRestartLock.lock();
+ if (mQuit) break;
+ }
+ }
+
+ mStop.clear();
+ mRestart.clear();
+ mCondition.signal();
+ mRestartLock.unlock();
+ ALOGV("goodbye");
+ return 0;
+}
+
+void SoundPool::quit()
+{
+ mRestartLock.lock();
+ mQuit = true;
+ mCondition.signal();
+ mCondition.wait(mRestartLock);
+ ALOGV("return from quit");
+ mRestartLock.unlock();
+}
+
+bool SoundPool::startThreads()
+{
+ createThreadEtc(beginThread, this, "SoundPool");
+ if (mDecodeThread == NULL)
+ mDecodeThread = new SoundPoolThread(this);
+ return mDecodeThread != NULL;
+}
+
+SoundChannel* SoundPool::findChannel(int channelID)
+{
+ for (int i = 0; i < mMaxChannels; ++i) {
+ if (mChannelPool[i].channelID() == channelID) {
+ return &mChannelPool[i];
+ }
+ }
+ return NULL;
+}
+
+SoundChannel* SoundPool::findNextChannel(int channelID)
+{
+ for (int i = 0; i < mMaxChannels; ++i) {
+ if (mChannelPool[i].nextChannelID() == channelID) {
+ return &mChannelPool[i];
+ }
+ }
+ return NULL;
+}
+
+int SoundPool::load(const char* path, int priority)
+{
+ ALOGV("load: path=%s, priority=%d", path, priority);
+ Mutex::Autolock lock(&mLock);
+ sp<Sample> sample = new Sample(++mNextSampleID, path);
+ mSamples.add(sample->sampleID(), sample);
+ doLoad(sample);
+ return sample->sampleID();
+}
+
+int SoundPool::load(int fd, int64_t offset, int64_t length, int priority)
+{
+ ALOGV("load: fd=%d, offset=%lld, length=%lld, priority=%d",
+ fd, offset, length, priority);
+ Mutex::Autolock lock(&mLock);
+ sp<Sample> sample = new Sample(++mNextSampleID, fd, offset, length);
+ mSamples.add(sample->sampleID(), sample);
+ doLoad(sample);
+ return sample->sampleID();
+}
+
+void SoundPool::doLoad(sp<Sample>& sample)
+{
+ ALOGV("doLoad: loading sample sampleID=%d", sample->sampleID());
+ sample->startLoad();
+ mDecodeThread->loadSample(sample->sampleID());
+}
+
+bool SoundPool::unload(int sampleID)
+{
+ ALOGV("unload: sampleID=%d", sampleID);
+ Mutex::Autolock lock(&mLock);
+ return mSamples.removeItem(sampleID);
+}
+
+int SoundPool::play(int sampleID, float leftVolume, float rightVolume,
+ int priority, int loop, float rate)
+{
+ ALOGV("play sampleID=%d, leftVolume=%f, rightVolume=%f, priority=%d, loop=%d, rate=%f",
+ sampleID, leftVolume, rightVolume, priority, loop, rate);
+ sp<Sample> sample;
+ SoundChannel* channel;
+ int channelID;
+
+ Mutex::Autolock lock(&mLock);
+
+ if (mQuit) {
+ return 0;
+ }
+ // is sample ready?
+ sample = findSample(sampleID);
+ if ((sample == 0) || (sample->state() != Sample::READY)) {
+ ALOGW(" sample %d not READY", sampleID);
+ return 0;
+ }
+
+ dump();
+
+ // allocate a channel
+ channel = allocateChannel_l(priority);
+
+ // no channel allocated - return 0
+ if (!channel) {
+ ALOGV("No channel allocated");
+ return 0;
+ }
+
+ channelID = ++mNextChannelID;
+
+ ALOGV("play channel %p state = %d", channel, channel->state());
+ channel->play(sample, channelID, leftVolume, rightVolume, priority, loop, rate);
+ return channelID;
+}
+
+SoundChannel* SoundPool::allocateChannel_l(int priority)
+{
+ List<SoundChannel*>::iterator iter;
+ SoundChannel* channel = NULL;
+
+ // allocate a channel
+ if (!mChannels.empty()) {
+ iter = mChannels.begin();
+ if (priority >= (*iter)->priority()) {
+ channel = *iter;
+ mChannels.erase(iter);
+ ALOGV("Allocated active channel");
+ }
+ }
+
+ // update priority and put it back in the list
+ if (channel) {
+ channel->setPriority(priority);
+ for (iter = mChannels.begin(); iter != mChannels.end(); ++iter) {
+ if (priority < (*iter)->priority()) {
+ break;
+ }
+ }
+ mChannels.insert(iter, channel);
+ }
+ return channel;
+}
+
+// move a channel from its current position to the front of the list
+void SoundPool::moveToFront_l(SoundChannel* channel)
+{
+ for (List<SoundChannel*>::iterator iter = mChannels.begin(); iter != mChannels.end(); ++iter) {
+ if (*iter == channel) {
+ mChannels.erase(iter);
+ mChannels.push_front(channel);
+ break;
+ }
+ }
+}
+
+void SoundPool::pause(int channelID)
+{
+ ALOGV("pause(%d)", channelID);
+ Mutex::Autolock lock(&mLock);
+ SoundChannel* channel = findChannel(channelID);
+ if (channel) {
+ channel->pause();
+ }
+}
+
+void SoundPool::autoPause()
+{
+ ALOGV("autoPause()");
+ Mutex::Autolock lock(&mLock);
+ for (int i = 0; i < mMaxChannels; ++i) {
+ SoundChannel* channel = &mChannelPool[i];
+ channel->autoPause();
+ }
+}
+
+void SoundPool::resume(int channelID)
+{
+ ALOGV("resume(%d)", channelID);
+ Mutex::Autolock lock(&mLock);
+ SoundChannel* channel = findChannel(channelID);
+ if (channel) {
+ channel->resume();
+ }
+}
+
+void SoundPool::autoResume()
+{
+ ALOGV("autoResume()");
+ Mutex::Autolock lock(&mLock);
+ for (int i = 0; i < mMaxChannels; ++i) {
+ SoundChannel* channel = &mChannelPool[i];
+ channel->autoResume();
+ }
+}
+
+void SoundPool::stop(int channelID)
+{
+ ALOGV("stop(%d)", channelID);
+ Mutex::Autolock lock(&mLock);
+ SoundChannel* channel = findChannel(channelID);
+ if (channel) {
+ channel->stop();
+ } else {
+ channel = findNextChannel(channelID);
+ if (channel)
+ channel->clearNextEvent();
+ }
+}
+
+void SoundPool::setVolume(int channelID, float leftVolume, float rightVolume)
+{
+ Mutex::Autolock lock(&mLock);
+ SoundChannel* channel = findChannel(channelID);
+ if (channel) {
+ channel->setVolume(leftVolume, rightVolume);
+ }
+}
+
+void SoundPool::setPriority(int channelID, int priority)
+{
+ ALOGV("setPriority(%d, %d)", channelID, priority);
+ Mutex::Autolock lock(&mLock);
+ SoundChannel* channel = findChannel(channelID);
+ if (channel) {
+ channel->setPriority(priority);
+ }
+}
+
+void SoundPool::setLoop(int channelID, int loop)
+{
+ ALOGV("setLoop(%d, %d)", channelID, loop);
+ Mutex::Autolock lock(&mLock);
+ SoundChannel* channel = findChannel(channelID);
+ if (channel) {
+ channel->setLoop(loop);
+ }
+}
+
+void SoundPool::setRate(int channelID, float rate)
+{
+ ALOGV("setRate(%d, %f)", channelID, rate);
+ Mutex::Autolock lock(&mLock);
+ SoundChannel* channel = findChannel(channelID);
+ if (channel) {
+ channel->setRate(rate);
+ }
+}
+
+// call with lock held
+void SoundPool::done_l(SoundChannel* channel)
+{
+ ALOGV("done_l(%d)", channel->channelID());
+ // if "stolen", play next event
+ if (channel->nextChannelID() != 0) {
+ ALOGV("add to restart list");
+ addToRestartList(channel);
+ }
+
+ // return to idle state
+ else {
+ ALOGV("move to front");
+ moveToFront_l(channel);
+ }
+}
+
+void SoundPool::setCallback(SoundPoolCallback* callback, void* user)
+{
+ Mutex::Autolock lock(&mCallbackLock);
+ mCallback = callback;
+ mUserData = user;
+}
+
+void SoundPool::notify(SoundPoolEvent event)
+{
+ Mutex::Autolock lock(&mCallbackLock);
+ if (mCallback != NULL) {
+ mCallback(event, this, mUserData);
+ }
+}
+
+void SoundPool::dump()
+{
+ for (int i = 0; i < mMaxChannels; ++i) {
+ mChannelPool[i].dump();
+ }
+}
+
+
+Sample::Sample(int sampleID, const char* url)
+{
+ init();
+ mSampleID = sampleID;
+ mUrl = strdup(url);
+ ALOGV("create sampleID=%d, url=%s", mSampleID, mUrl);
+}
+
+Sample::Sample(int sampleID, int fd, int64_t offset, int64_t length)
+{
+ init();
+ mSampleID = sampleID;
+ mFd = dup(fd);
+ mOffset = offset;
+ mLength = length;
+ ALOGV("create sampleID=%d, fd=%d, offset=%lld, length=%lld", mSampleID, mFd, mLength, mOffset);
+}
+
+void Sample::init()
+{
+ mData = 0;
+ mSize = 0;
+ mRefCount = 0;
+ mSampleID = 0;
+ mState = UNLOADED;
+ mFd = -1;
+ mOffset = 0;
+ mLength = 0;
+ mUrl = 0;
+}
+
+Sample::~Sample()
+{
+ ALOGV("Sample::destructor sampleID=%d, fd=%d", mSampleID, mFd);
+ if (mFd > 0) {
+ ALOGV("close(%d)", mFd);
+ ::close(mFd);
+ }
+ mData.clear();
+ delete mUrl;
+}
+
+status_t Sample::doLoad()
+{
+ uint32_t sampleRate;
+ int numChannels;
+ audio_format_t format;
+ sp<IMemory> p;
+ ALOGV("Start decode");
+ if (mUrl) {
+ p = MediaPlayer::decode(mUrl, &sampleRate, &numChannels, &format);
+ } else {
+ p = MediaPlayer::decode(mFd, mOffset, mLength, &sampleRate, &numChannels, &format);
+ ALOGV("close(%d)", mFd);
+ ::close(mFd);
+ mFd = -1;
+ }
+ if (p == 0) {
+ ALOGE("Unable to load sample: %s", mUrl);
+ return -1;
+ }
+ ALOGV("pointer = %p, size = %u, sampleRate = %u, numChannels = %d",
+ p->pointer(), p->size(), sampleRate, numChannels);
+
+ if (sampleRate > kMaxSampleRate) {
+ ALOGE("Sample rate (%u) out of range", sampleRate);
+ return - 1;
+ }
+
+ if ((numChannels < 1) || (numChannels > 2)) {
+ ALOGE("Sample channel count (%d) out of range", numChannels);
+ return - 1;
+ }
+
+ //_dumpBuffer(p->pointer(), p->size());
+ uint8_t* q = static_cast<uint8_t*>(p->pointer()) + p->size() - 10;
+ //_dumpBuffer(q, 10, 10, false);
+
+ mData = p;
+ mSize = p->size();
+ mSampleRate = sampleRate;
+ mNumChannels = numChannels;
+ mFormat = format;
+ mState = READY;
+ return 0;
+}
+
+
+void SoundChannel::init(SoundPool* soundPool)
+{
+ mSoundPool = soundPool;
+}
+
+// call with sound pool lock held
+void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftVolume,
+ float rightVolume, int priority, int loop, float rate)
+{
+ AudioTrack* oldTrack;
+ AudioTrack* newTrack;
+ status_t status;
+
+ { // scope for the lock
+ Mutex::Autolock lock(&mLock);
+
+ ALOGV("SoundChannel::play %p: sampleID=%d, channelID=%d, leftVolume=%f, rightVolume=%f,"
+ " priority=%d, loop=%d, rate=%f",
+ this, sample->sampleID(), nextChannelID, leftVolume, rightVolume,
+ priority, loop, rate);
+
+ // if not idle, this voice is being stolen
+ if (mState != IDLE) {
+ ALOGV("channel %d stolen - event queued for channel %d", channelID(), nextChannelID);
+ mNextEvent.set(sample, nextChannelID, leftVolume, rightVolume, priority, loop, rate);
+ stop_l();
+ return;
+ }
+
+ // initialize track
+ int afFrameCount;
+ int afSampleRate;
+ audio_stream_type_t streamType = mSoundPool->streamType();
+ if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
+ afFrameCount = kDefaultFrameCount;
+ }
+ if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
+ afSampleRate = kDefaultSampleRate;
+ }
+ int numChannels = sample->numChannels();
+ uint32_t sampleRate = uint32_t(float(sample->sampleRate()) * rate + 0.5);
+ uint32_t totalFrames = (kDefaultBufferCount * afFrameCount * sampleRate) / afSampleRate;
+ uint32_t bufferFrames = (totalFrames + (kDefaultBufferCount - 1)) / kDefaultBufferCount;
+ uint32_t frameCount = 0;
+
+ if (loop) {
+ frameCount = sample->size()/numChannels/
+ ((sample->format() == AUDIO_FORMAT_PCM_16_BIT) ? sizeof(int16_t) : sizeof(uint8_t));
+ }
+
+#ifndef USE_SHARED_MEM_BUFFER
+ // Ensure minimum audio buffer size in case of short looped sample
+ if(frameCount < totalFrames) {
+ frameCount = totalFrames;
+ }
+#endif
+
+ // mToggle toggles each time a track is started on a given channel.
+ // The toggle is concatenated with the SoundChannel address and passed to AudioTrack
+ // as callback user data. This enables the detection of callbacks received from the old
+ // audio track while the new one is being started and avoids processing them with
+ // wrong audio audio buffer size (mAudioBufferSize)
+ unsigned long toggle = mToggle ^ 1;
+ void *userData = (void *)((unsigned long)this | toggle);
+ uint32_t channels = (numChannels == 2) ?
+ AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO;
+
+ // do not create a new audio track if current track is compatible with sample parameters
+#ifdef USE_SHARED_MEM_BUFFER
+ newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
+ channels, sample->getIMemory(), AUDIO_POLICY_OUTPUT_FLAG_NONE, callback, userData);
+#else
+ newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
+ channels, frameCount, AUDIO_POLICY_OUTPUT_FLAG_NONE, callback, userData,
+ bufferFrames);
+#endif
+ oldTrack = mAudioTrack;
+ status = newTrack->initCheck();
+ if (status != NO_ERROR) {
+ ALOGE("Error creating AudioTrack");
+ goto exit;
+ }
+ ALOGV("setVolume %p", newTrack);
+ newTrack->setVolume(leftVolume, rightVolume);
+ newTrack->setLoop(0, frameCount, loop);
+
+ // From now on, AudioTrack callbacks received with previous toggle value will be ignored.
+ mToggle = toggle;
+ mAudioTrack = newTrack;
+ mPos = 0;
+ mSample = sample;
+ mChannelID = nextChannelID;
+ mPriority = priority;
+ mLoop = loop;
+ mLeftVolume = leftVolume;
+ mRightVolume = rightVolume;
+ mNumChannels = numChannels;
+ mRate = rate;
+ clearNextEvent();
+ mState = PLAYING;
+ mAudioTrack->start();
+ mAudioBufferSize = newTrack->frameCount()*newTrack->frameSize();
+ }
+
+exit:
+ ALOGV("delete oldTrack %p", oldTrack);
+ delete oldTrack;
+ if (status != NO_ERROR) {
+ delete newTrack;
+ mAudioTrack = NULL;
+ }
+}
+
+void SoundChannel::nextEvent()
+{
+ sp<Sample> sample;
+ int nextChannelID;
+ float leftVolume;
+ float rightVolume;
+ int priority;
+ int loop;
+ float rate;
+
+ // check for valid event
+ {
+ Mutex::Autolock lock(&mLock);
+ nextChannelID = mNextEvent.channelID();
+ if (nextChannelID == 0) {
+ ALOGV("stolen channel has no event");
+ return;
+ }
+
+ sample = mNextEvent.sample();
+ leftVolume = mNextEvent.leftVolume();
+ rightVolume = mNextEvent.rightVolume();
+ priority = mNextEvent.priority();
+ loop = mNextEvent.loop();
+ rate = mNextEvent.rate();
+ }
+
+ ALOGV("Starting stolen channel %d -> %d", channelID(), nextChannelID);
+ play(sample, nextChannelID, leftVolume, rightVolume, priority, loop, rate);
+}
+
+void SoundChannel::callback(int event, void* user, void *info)
+{
+ SoundChannel* channel = static_cast<SoundChannel*>((void *)((unsigned long)user & ~1));
+
+ channel->process(event, info, (unsigned long)user & 1);
+}
+
+void SoundChannel::process(int event, void *info, unsigned long toggle)
+{
+ //ALOGV("process(%d)", mChannelID);
+
+ Mutex::Autolock lock(&mLock);
+
+ AudioTrack::Buffer* b = NULL;
+ if (event == AudioTrack::EVENT_MORE_DATA) {
+ b = static_cast<AudioTrack::Buffer *>(info);
+ }
+
+ if (mToggle != toggle) {
+ ALOGV("process wrong toggle %p channel %d", this, mChannelID);
+ if (b != NULL) {
+ b->size = 0;
+ }
+ return;
+ }
+
+ sp<Sample> sample = mSample;
+
+// ALOGV("SoundChannel::process event %d", event);
+
+ if (event == AudioTrack::EVENT_MORE_DATA) {
+
+ // check for stop state
+ if (b->size == 0) return;
+
+ if (mState == IDLE) {
+ b->size = 0;
+ return;
+ }
+
+ if (sample != 0) {
+ // fill buffer
+ uint8_t* q = (uint8_t*) b->i8;
+ size_t count = 0;
+
+ if (mPos < (int)sample->size()) {
+ uint8_t* p = sample->data() + mPos;
+ count = sample->size() - mPos;
+ if (count > b->size) {
+ count = b->size;
+ }
+ memcpy(q, p, count);
+// ALOGV("fill: q=%p, p=%p, mPos=%u, b->size=%u, count=%d", q, p, mPos, b->size, count);
+ } else if (mPos < mAudioBufferSize) {
+ count = mAudioBufferSize - mPos;
+ if (count > b->size) {
+ count = b->size;
+ }
+ memset(q, 0, count);
+// ALOGV("fill extra: q=%p, mPos=%u, b->size=%u, count=%d", q, mPos, b->size, count);
+ }
+
+ mPos += count;
+ b->size = count;
+ //ALOGV("buffer=%p, [0]=%d", b->i16, b->i16[0]);
+ }
+ } else if (event == AudioTrack::EVENT_UNDERRUN) {
+ ALOGV("process %p channel %d EVENT_UNDERRUN", this, mChannelID);
+ mSoundPool->addToStopList(this);
+ } else if (event == AudioTrack::EVENT_LOOP_END) {
+ ALOGV("End loop %p channel %d count %d", this, mChannelID, *(int *)info);
+ }
+}
+
+
+// call with lock held
+bool SoundChannel::doStop_l()
+{
+ if (mState != IDLE) {
+ setVolume_l(0, 0);
+ ALOGV("stop");
+ mAudioTrack->stop();
+ mSample.clear();
+ mState = IDLE;
+ mPriority = IDLE_PRIORITY;
+ return true;
+ }
+ return false;
+}
+
+// call with lock held and sound pool lock held
+void SoundChannel::stop_l()
+{
+ if (doStop_l()) {
+ mSoundPool->done_l(this);
+ }
+}
+
+// call with sound pool lock held
+void SoundChannel::stop()
+{
+ bool stopped;
+ {
+ Mutex::Autolock lock(&mLock);
+ stopped = doStop_l();
+ }
+
+ if (stopped) {
+ mSoundPool->done_l(this);
+ }
+}
+
+//FIXME: Pause is a little broken right now
+void SoundChannel::pause()
+{
+ Mutex::Autolock lock(&mLock);
+ if (mState == PLAYING) {
+ ALOGV("pause track");
+ mState = PAUSED;
+ mAudioTrack->pause();
+ }
+}
+
+void SoundChannel::autoPause()
+{
+ Mutex::Autolock lock(&mLock);
+ if (mState == PLAYING) {
+ ALOGV("pause track");
+ mState = PAUSED;
+ mAutoPaused = true;
+ mAudioTrack->pause();
+ }
+}
+
+void SoundChannel::resume()
+{
+ Mutex::Autolock lock(&mLock);
+ if (mState == PAUSED) {
+ ALOGV("resume track");
+ mState = PLAYING;
+ mAutoPaused = false;
+ mAudioTrack->start();
+ }
+}
+
+void SoundChannel::autoResume()
+{
+ Mutex::Autolock lock(&mLock);
+ if (mAutoPaused && (mState == PAUSED)) {
+ ALOGV("resume track");
+ mState = PLAYING;
+ mAutoPaused = false;
+ mAudioTrack->start();
+ }
+}
+
+void SoundChannel::setRate(float rate)
+{
+ Mutex::Autolock lock(&mLock);
+ if (mAudioTrack != NULL && mSample != 0) {
+ uint32_t sampleRate = uint32_t(float(mSample->sampleRate()) * rate + 0.5);
+ mAudioTrack->setSampleRate(sampleRate);
+ mRate = rate;
+ }
+}
+
+// call with lock held
+void SoundChannel::setVolume_l(float leftVolume, float rightVolume)
+{
+ mLeftVolume = leftVolume;
+ mRightVolume = rightVolume;
+ if (mAudioTrack != NULL)
+ mAudioTrack->setVolume(leftVolume, rightVolume);
+}
+
+void SoundChannel::setVolume(float leftVolume, float rightVolume)
+{
+ Mutex::Autolock lock(&mLock);
+ setVolume_l(leftVolume, rightVolume);
+}
+
+void SoundChannel::setLoop(int loop)
+{
+ Mutex::Autolock lock(&mLock);
+ if (mAudioTrack != NULL && mSample != 0) {
+ uint32_t loopEnd = mSample->size()/mNumChannels/
+ ((mSample->format() == AUDIO_FORMAT_PCM_16_BIT) ? sizeof(int16_t) : sizeof(uint8_t));
+ mAudioTrack->setLoop(0, loopEnd, loop);
+ mLoop = loop;
+ }
+}
+
+SoundChannel::~SoundChannel()
+{
+ ALOGV("SoundChannel destructor %p", this);
+ {
+ Mutex::Autolock lock(&mLock);
+ clearNextEvent();
+ doStop_l();
+ }
+ // do not call AudioTrack destructor with mLock held as it will wait for the AudioTrack
+ // callback thread to exit which may need to execute process() and acquire the mLock.
+ delete mAudioTrack;
+}
+
+void SoundChannel::dump()
+{
+ ALOGV("mState = %d mChannelID=%d, mNumChannels=%d, mPos = %d, mPriority=%d, mLoop=%d",
+ mState, mChannelID, mNumChannels, mPos, mPriority, mLoop);
+}
+
+void SoundEvent::set(const sp<Sample>& sample, int channelID, float leftVolume,
+ float rightVolume, int priority, int loop, float rate)
+{
+ mSample = sample;
+ mChannelID = channelID;
+ mLeftVolume = leftVolume;
+ mRightVolume = rightVolume;
+ mPriority = priority;
+ mLoop = loop;
+ mRate =rate;
+}
+
+} // end namespace android