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authorMarco Nelissen <marcone@google.com>2014-12-04 08:57:56 -0800
committerMarco Nelissen <marcone@google.com>2015-01-26 07:57:39 -0800
commitb7848f1c442bc8c71020957948d1b003a62a25b7 (patch)
treeb0a1dfd418429171c0af5379a1820569775a840b
parent1ed13b3b443735e7f7b5002954f31741b0b1c572 (diff)
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Remove MediaPlayerService::decode()
and make SoundPool use MediaCodec for decoding files to PCM. Bug: 18239054 Change-Id: Ia144fc1bbb0d2787638ee972e2224339b4965310
-rw-r--r--include/media/IMediaPlayerService.h11
-rw-r--r--include/media/MediaPlayerInterface.h1
-rw-r--r--include/media/SoundPool.h241
-rw-r--r--include/media/mediaplayer.h11
-rw-r--r--media/libmedia/Android.mk2
-rw-r--r--media/libmedia/IMediaPlayerService.cpp107
-rw-r--r--media/libmedia/SoundPool.cpp921
-rw-r--r--media/libmedia/SoundPoolThread.cpp114
-rw-r--r--media/libmedia/SoundPoolThread.h66
-rw-r--r--media/libmedia/mediaplayer.cpp41
-rw-r--r--media/libmediaplayerservice/MediaPlayerService.cpp317
-rw-r--r--media/libmediaplayerservice/MediaPlayerService.h83
-rw-r--r--media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp13
-rw-r--r--media/libstagefright/AwesomePlayer.cpp5
14 files changed, 13 insertions, 1920 deletions
diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h
index d7e584a..67b599a 100644
--- a/include/media/IMediaPlayerService.h
+++ b/include/media/IMediaPlayerService.h
@@ -51,17 +51,6 @@ public:
virtual sp<IMediaMetadataRetriever> createMetadataRetriever() = 0;
virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0) = 0;
- virtual status_t decode(
- const sp<IMediaHTTPService> &httpService,
- const char* url,
- uint32_t *pSampleRate,
- int* pNumChannels,
- audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
-
- virtual status_t decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate,
- int* pNumChannels, audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap, size_t *pSize) = 0;
virtual sp<IOMX> getOMX() = 0;
virtual sp<ICrypto> makeCrypto() = 0;
virtual sp<IDrm> makeDrm() = 0;
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index 0f38c16..b980421 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -88,7 +88,6 @@ public:
virtual ~AudioSink() {}
virtual bool ready() const = 0; // audio output is open and ready
- virtual bool realtime() const = 0; // audio output is real-time output
virtual ssize_t bufferSize() const = 0;
virtual ssize_t frameCount() const = 0;
virtual ssize_t channelCount() const = 0;
diff --git a/include/media/SoundPool.h b/include/media/SoundPool.h
deleted file mode 100644
index 5830475..0000000
--- a/include/media/SoundPool.h
+++ /dev/null
@@ -1,241 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef SOUNDPOOL_H_
-#define SOUNDPOOL_H_
-
-#include <utils/threads.h>
-#include <utils/List.h>
-#include <utils/Vector.h>
-#include <utils/KeyedVector.h>
-#include <media/AudioTrack.h>
-#include <binder/MemoryHeapBase.h>
-#include <binder/MemoryBase.h>
-
-namespace android {
-
-static const int IDLE_PRIORITY = -1;
-
-// forward declarations
-class SoundEvent;
-class SoundPoolThread;
-class SoundPool;
-
-// for queued events
-class SoundPoolEvent {
-public:
- SoundPoolEvent(int msg, int arg1=0, int arg2=0) :
- mMsg(msg), mArg1(arg1), mArg2(arg2) {}
- int mMsg;
- int mArg1;
- int mArg2;
- enum MessageType { INVALID, SAMPLE_LOADED };
-};
-
-// callback function prototype
-typedef void SoundPoolCallback(SoundPoolEvent event, SoundPool* soundPool, void* user);
-
-// tracks samples used by application
-class Sample : public RefBase {
-public:
- enum sample_state { UNLOADED, LOADING, READY, UNLOADING };
- Sample(int sampleID, const char* url);
- Sample(int sampleID, int fd, int64_t offset, int64_t length);
- ~Sample();
- int sampleID() { return mSampleID; }
- int numChannels() { return mNumChannels; }
- int sampleRate() { return mSampleRate; }
- audio_format_t format() { return mFormat; }
- size_t size() { return mSize; }
- int state() { return mState; }
- uint8_t* data() { return static_cast<uint8_t*>(mData->pointer()); }
- status_t doLoad();
- void startLoad() { mState = LOADING; }
- sp<IMemory> getIMemory() { return mData; }
-
- // hack
- void init(int numChannels, int sampleRate, audio_format_t format, size_t size,
- sp<IMemory> data ) {
- mNumChannels = numChannels; mSampleRate = sampleRate; mFormat = format; mSize = size;
- mData = data; }
-
-private:
- void init();
-
- size_t mSize;
- volatile int32_t mRefCount;
- uint16_t mSampleID;
- uint16_t mSampleRate;
- uint8_t mState : 3;
- uint8_t mNumChannels : 2;
- audio_format_t mFormat;
- int mFd;
- int64_t mOffset;
- int64_t mLength;
- char* mUrl;
- sp<IMemory> mData;
- sp<MemoryHeapBase> mHeap;
-};
-
-// stores pending events for stolen channels
-class SoundEvent
-{
-public:
- SoundEvent() : mChannelID(0), mLeftVolume(0), mRightVolume(0),
- mPriority(IDLE_PRIORITY), mLoop(0), mRate(0) {}
- void set(const sp<Sample>& sample, int channelID, float leftVolume,
- float rightVolume, int priority, int loop, float rate);
- sp<Sample> sample() { return mSample; }
- int channelID() { return mChannelID; }
- float leftVolume() { return mLeftVolume; }
- float rightVolume() { return mRightVolume; }
- int priority() { return mPriority; }
- int loop() { return mLoop; }
- float rate() { return mRate; }
- void clear() { mChannelID = 0; mSample.clear(); }
-
-protected:
- sp<Sample> mSample;
- int mChannelID;
- float mLeftVolume;
- float mRightVolume;
- int mPriority;
- int mLoop;
- float mRate;
-};
-
-// for channels aka AudioTracks
-class SoundChannel : public SoundEvent {
-public:
- enum state { IDLE, RESUMING, STOPPING, PAUSED, PLAYING };
- SoundChannel() : mState(IDLE), mNumChannels(1),
- mPos(0), mToggle(0), mAutoPaused(false) {}
- ~SoundChannel();
- void init(SoundPool* soundPool);
- void play(const sp<Sample>& sample, int channelID, float leftVolume, float rightVolume,
- int priority, int loop, float rate);
- void setVolume_l(float leftVolume, float rightVolume);
- void setVolume(float leftVolume, float rightVolume);
- void stop_l();
- void stop();
- void pause();
- void autoPause();
- void resume();
- void autoResume();
- void setRate(float rate);
- int state() { return mState; }
- void setPriority(int priority) { mPriority = priority; }
- void setLoop(int loop);
- int numChannels() { return mNumChannels; }
- void clearNextEvent() { mNextEvent.clear(); }
- void nextEvent();
- int nextChannelID() { return mNextEvent.channelID(); }
- void dump();
-
-private:
- static void callback(int event, void* user, void *info);
- void process(int event, void *info, unsigned long toggle);
- bool doStop_l();
-
- SoundPool* mSoundPool;
- sp<AudioTrack> mAudioTrack;
- SoundEvent mNextEvent;
- Mutex mLock;
- int mState;
- int mNumChannels;
- int mPos;
- int mAudioBufferSize;
- unsigned long mToggle;
- bool mAutoPaused;
-};
-
-// application object for managing a pool of sounds
-class SoundPool {
- friend class SoundPoolThread;
- friend class SoundChannel;
-public:
- SoundPool(int maxChannels, const audio_attributes_t* pAttributes);
- ~SoundPool();
- int load(const char* url, int priority);
- int load(int fd, int64_t offset, int64_t length, int priority);
- bool unload(int sampleID);
- int play(int sampleID, float leftVolume, float rightVolume, int priority,
- int loop, float rate);
- void pause(int channelID);
- void autoPause();
- void resume(int channelID);
- void autoResume();
- void stop(int channelID);
- void setVolume(int channelID, float leftVolume, float rightVolume);
- void setPriority(int channelID, int priority);
- void setLoop(int channelID, int loop);
- void setRate(int channelID, float rate);
- const audio_attributes_t* attributes() { return &mAttributes; }
-
- // called from SoundPoolThread
- void sampleLoaded(int sampleID);
-
- // called from AudioTrack thread
- void done_l(SoundChannel* channel);
-
- // callback function
- void setCallback(SoundPoolCallback* callback, void* user);
- void* getUserData() { return mUserData; }
-
-private:
- SoundPool() {} // no default constructor
- bool startThreads();
- void doLoad(sp<Sample>& sample);
- sp<Sample> findSample(int sampleID) { return mSamples.valueFor(sampleID); }
- SoundChannel* findChannel (int channelID);
- SoundChannel* findNextChannel (int channelID);
- SoundChannel* allocateChannel_l(int priority);
- void moveToFront_l(SoundChannel* channel);
- void notify(SoundPoolEvent event);
- void dump();
-
- // restart thread
- void addToRestartList(SoundChannel* channel);
- void addToStopList(SoundChannel* channel);
- static int beginThread(void* arg);
- int run();
- void quit();
-
- Mutex mLock;
- Mutex mRestartLock;
- Condition mCondition;
- SoundPoolThread* mDecodeThread;
- SoundChannel* mChannelPool;
- List<SoundChannel*> mChannels;
- List<SoundChannel*> mRestart;
- List<SoundChannel*> mStop;
- DefaultKeyedVector< int, sp<Sample> > mSamples;
- int mMaxChannels;
- audio_attributes_t mAttributes;
- int mAllocated;
- int mNextSampleID;
- int mNextChannelID;
- bool mQuit;
-
- // callback
- Mutex mCallbackLock;
- SoundPoolCallback* mCallback;
- void* mUserData;
-};
-
-} // end namespace android
-
-#endif /*SOUNDPOOL_H_*/
diff --git a/include/media/mediaplayer.h b/include/media/mediaplayer.h
index 9cc208e..5830933 100644
--- a/include/media/mediaplayer.h
+++ b/include/media/mediaplayer.h
@@ -232,17 +232,6 @@ public:
bool isLooping();
status_t setVolume(float leftVolume, float rightVolume);
void notify(int msg, int ext1, int ext2, const Parcel *obj = NULL);
- static status_t decode(
- const sp<IMediaHTTPService> &httpService,
- const char* url,
- uint32_t *pSampleRate,
- int* pNumChannels,
- audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap,
- size_t *pSize);
- static status_t decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate,
- int* pNumChannels, audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap, size_t *pSize);
status_t invoke(const Parcel& request, Parcel *reply);
status_t setMetadataFilter(const Parcel& filter);
status_t getMetadata(bool update_only, bool apply_filter, Parcel *metadata);
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index df6eb57..b91b7d1 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -58,8 +58,6 @@ LOCAL_SRC_FILES:= \
AudioEffect.cpp \
Visualizer.cpp \
MemoryLeakTrackUtil.cpp \
- SoundPool.cpp \
- SoundPoolThread.cpp \
StringArray.cpp
LOCAL_SRC_FILES += ../libnbaio/roundup.c
diff --git a/media/libmedia/IMediaPlayerService.cpp b/media/libmedia/IMediaPlayerService.cpp
index 4d33d40..feea267 100644
--- a/media/libmedia/IMediaPlayerService.cpp
+++ b/media/libmedia/IMediaPlayerService.cpp
@@ -39,8 +39,6 @@ namespace android {
enum {
CREATE = IBinder::FIRST_CALL_TRANSACTION,
- DECODE_URL,
- DECODE_FD,
CREATE_MEDIA_RECORDER,
CREATE_METADATA_RETRIEVER,
GET_OMX,
@@ -88,59 +86,6 @@ public:
return interface_cast<IMediaRecorder>(reply.readStrongBinder());
}
- virtual status_t decode(
- const sp<IMediaHTTPService> &httpService,
- const char* url,
- uint32_t *pSampleRate,
- int* pNumChannels,
- audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap,
- size_t *pSize)
- {
- Parcel data, reply;
- data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
- data.writeInt32(httpService != NULL);
- if (httpService != NULL) {
- data.writeStrongBinder(IInterface::asBinder(httpService));
- }
- data.writeCString(url);
- data.writeStrongBinder(IInterface::asBinder(heap));
- status_t status = remote()->transact(DECODE_URL, data, &reply);
- if (status == NO_ERROR) {
- status = (status_t)reply.readInt32();
- if (status == NO_ERROR) {
- *pSampleRate = uint32_t(reply.readInt32());
- *pNumChannels = reply.readInt32();
- *pFormat = (audio_format_t)reply.readInt32();
- *pSize = (size_t)reply.readInt32();
- }
- }
- return status;
- }
-
- virtual status_t decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate,
- int* pNumChannels, audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap, size_t *pSize)
- {
- Parcel data, reply;
- data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
- data.writeFileDescriptor(fd);
- data.writeInt64(offset);
- data.writeInt64(length);
- data.writeStrongBinder(IInterface::asBinder(heap));
- status_t status = remote()->transact(DECODE_FD, data, &reply);
- if (status == NO_ERROR) {
- status = (status_t)reply.readInt32();
- if (status == NO_ERROR) {
- *pSampleRate = uint32_t(reply.readInt32());
- *pNumChannels = reply.readInt32();
- *pFormat = (audio_format_t)reply.readInt32();
- *pSize = (size_t)reply.readInt32();
- }
- }
- return status;
- }
-
virtual sp<IOMX> getOMX() {
Parcel data, reply;
data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
@@ -219,58 +164,6 @@ status_t BnMediaPlayerService::onTransact(
reply->writeStrongBinder(IInterface::asBinder(player));
return NO_ERROR;
} break;
- case DECODE_URL: {
- CHECK_INTERFACE(IMediaPlayerService, data, reply);
- sp<IMediaHTTPService> httpService;
- if (data.readInt32()) {
- httpService =
- interface_cast<IMediaHTTPService>(data.readStrongBinder());
- }
- const char* url = data.readCString();
- sp<IMemoryHeap> heap = interface_cast<IMemoryHeap>(data.readStrongBinder());
- uint32_t sampleRate;
- int numChannels;
- audio_format_t format;
- size_t size;
- status_t status =
- decode(httpService,
- url,
- &sampleRate,
- &numChannels,
- &format,
- heap,
- &size);
- reply->writeInt32(status);
- if (status == NO_ERROR) {
- reply->writeInt32(sampleRate);
- reply->writeInt32(numChannels);
- reply->writeInt32((int32_t)format);
- reply->writeInt32((int32_t)size);
- }
- return NO_ERROR;
- } break;
- case DECODE_FD: {
- CHECK_INTERFACE(IMediaPlayerService, data, reply);
- int fd = dup(data.readFileDescriptor());
- int64_t offset = data.readInt64();
- int64_t length = data.readInt64();
- sp<IMemoryHeap> heap = interface_cast<IMemoryHeap>(data.readStrongBinder());
- uint32_t sampleRate;
- int numChannels;
- audio_format_t format;
- size_t size;
- status_t status = decode(fd, offset, length, &sampleRate, &numChannels, &format,
- heap, &size);
- ::close(fd);
- reply->writeInt32(status);
- if (status == NO_ERROR) {
- reply->writeInt32(sampleRate);
- reply->writeInt32(numChannels);
- reply->writeInt32((int32_t)format);
- reply->writeInt32((int32_t)size);
- }
- return NO_ERROR;
- } break;
case CREATE_MEDIA_RECORDER: {
CHECK_INTERFACE(IMediaPlayerService, data, reply);
sp<IMediaRecorder> recorder = createMediaRecorder();
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
deleted file mode 100644
index d2e381b..0000000
--- a/media/libmedia/SoundPool.cpp
+++ /dev/null
@@ -1,921 +0,0 @@
-/*
- * 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 <inttypes.h>
-
-#include <utils/Log.h>
-
-#define USE_SHARED_MEM_BUFFER
-
-#include <media/AudioTrack.h>
-#include <media/IMediaHTTPService.h>
-#include <media/mediaplayer.h>
-#include <media/SoundPool.h>
-#include "SoundPoolThread.h"
-#include <media/AudioPolicyHelper.h>
-
-namespace android
-{
-
-int kDefaultBufferCount = 4;
-uint32_t kMaxSampleRate = 48000;
-uint32_t kDefaultSampleRate = 44100;
-uint32_t kDefaultFrameCount = 1200;
-size_t kDefaultHeapSize = 1024 * 1024; // 1MB
-
-
-SoundPool::SoundPool(int maxChannels, const audio_attributes_t* pAttributes)
-{
- ALOGV("SoundPool constructor: maxChannels=%d, attr.usage=%d, attr.flags=0x%x, attr.tags=%s",
- maxChannels, pAttributes->usage, pAttributes->flags, pAttributes->tags);
-
- // 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;
- memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
- 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 __unused)
-{
- 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 __unused)
-{
- ALOGV("load: fd=%d, offset=%" PRId64 ", length=%" PRId64 ", 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=%" PRId64 " length=%" PRId64,
- mSampleID, mFd, mLength, mOffset);
-}
-
-void Sample::init()
-{
- 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);
- }
- free(mUrl);
-}
-
-status_t Sample::doLoad()
-{
- uint32_t sampleRate;
- int numChannels;
- audio_format_t format;
- status_t status;
- mHeap = new MemoryHeapBase(kDefaultHeapSize);
-
- ALOGV("Start decode");
- if (mUrl) {
- status = MediaPlayer::decode(
- NULL /* httpService */,
- mUrl,
- &sampleRate,
- &numChannels,
- &format,
- mHeap,
- &mSize);
- } else {
- status = MediaPlayer::decode(mFd, mOffset, mLength, &sampleRate, &numChannels, &format,
- mHeap, &mSize);
- ALOGV("close(%d)", mFd);
- ::close(mFd);
- mFd = -1;
- }
- if (status != NO_ERROR) {
- ALOGE("Unable to load sample: %s", mUrl);
- goto error;
- }
- ALOGV("pointer = %p, size = %zu, sampleRate = %u, numChannels = %d",
- mHeap->getBase(), mSize, sampleRate, numChannels);
-
- if (sampleRate > kMaxSampleRate) {
- ALOGE("Sample rate (%u) out of range", sampleRate);
- status = BAD_VALUE;
- goto error;
- }
-
- if ((numChannels < 1) || (numChannels > 2)) {
- ALOGE("Sample channel count (%d) out of range", numChannels);
- status = BAD_VALUE;
- goto error;
- }
-
- mData = new MemoryBase(mHeap, 0, mSize);
- mSampleRate = sampleRate;
- mNumChannels = numChannels;
- mFormat = format;
- mState = READY;
- return NO_ERROR;
-
-error:
- mHeap.clear();
- return status;
-}
-
-
-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)
-{
- sp<AudioTrack> oldTrack;
- sp<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
- size_t afFrameCount;
- uint32_t afSampleRate;
- audio_stream_type_t streamType = audio_attributes_to_stream_type(mSoundPool->attributes());
- 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;
- size_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);
- audio_channel_mask_t channelMask = audio_channel_out_mask_from_count(numChannels);
-
- // 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(),
- channelMask, sample->getIMemory(), AUDIO_OUTPUT_FLAG_FAST, callback, userData);
-#else
- newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
- channelMask, frameCount, AUDIO_OUTPUT_FLAG_FAST, callback, userData,
- bufferFrames);
-#endif
- oldTrack = mAudioTrack;
- status = newTrack->initCheck();
- if (status != NO_ERROR) {
- ALOGE("Error creating AudioTrack");
- goto exit;
- }
- ALOGV("setVolume %p", newTrack.get());
- 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.get());
- if (status != NO_ERROR) {
- mAudioTrack.clear();
- }
-}
-
-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 || event == AudioTrack::EVENT_BUFFER_END ||
- event == AudioTrack::EVENT_NEW_IAUDIOTRACK) {
- ALOGV("process %p channel %d event %s",
- this, mChannelID, (event == AudioTrack::EVENT_UNDERRUN) ? "UNDERRUN" :
- (event == AudioTrack::EVENT_BUFFER_END) ? "BUFFER_END" : "NEW_IAUDIOTRACK");
- mSoundPool->addToStopList(this);
- } else if (event == AudioTrack::EVENT_LOOP_END) {
- ALOGV("End loop %p channel %d", this, mChannelID);
- } else {
- ALOGW("SoundChannel::process unexpected event %d", event);
- }
-}
-
-
-// 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.
- mAudioTrack.clear();
-}
-
-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
diff --git a/media/libmedia/SoundPoolThread.cpp b/media/libmedia/SoundPoolThread.cpp
deleted file mode 100644
index ba3b482..0000000
--- a/media/libmedia/SoundPoolThread.cpp
+++ /dev/null
@@ -1,114 +0,0 @@
-/*
- * 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 "SoundPoolThread"
-#include "utils/Log.h"
-
-#include "SoundPoolThread.h"
-
-namespace android {
-
-void SoundPoolThread::write(SoundPoolMsg msg) {
- Mutex::Autolock lock(&mLock);
- while (mMsgQueue.size() >= maxMessages) {
- mCondition.wait(mLock);
- }
-
- // if thread is quitting, don't add to queue
- if (mRunning) {
- mMsgQueue.push(msg);
- mCondition.signal();
- }
-}
-
-const SoundPoolMsg SoundPoolThread::read() {
- Mutex::Autolock lock(&mLock);
- while (mMsgQueue.size() == 0) {
- mCondition.wait(mLock);
- }
- SoundPoolMsg msg = mMsgQueue[0];
- mMsgQueue.removeAt(0);
- mCondition.signal();
- return msg;
-}
-
-void SoundPoolThread::quit() {
- Mutex::Autolock lock(&mLock);
- if (mRunning) {
- mRunning = false;
- mMsgQueue.clear();
- mMsgQueue.push(SoundPoolMsg(SoundPoolMsg::KILL, 0));
- mCondition.signal();
- mCondition.wait(mLock);
- }
- ALOGV("return from quit");
-}
-
-SoundPoolThread::SoundPoolThread(SoundPool* soundPool) :
- mSoundPool(soundPool)
-{
- mMsgQueue.setCapacity(maxMessages);
- if (createThreadEtc(beginThread, this, "SoundPoolThread")) {
- mRunning = true;
- }
-}
-
-SoundPoolThread::~SoundPoolThread()
-{
- quit();
-}
-
-int SoundPoolThread::beginThread(void* arg) {
- ALOGV("beginThread");
- SoundPoolThread* soundPoolThread = (SoundPoolThread*)arg;
- return soundPoolThread->run();
-}
-
-int SoundPoolThread::run() {
- ALOGV("run");
- for (;;) {
- SoundPoolMsg msg = read();
- ALOGV("Got message m=%d, mData=%d", msg.mMessageType, msg.mData);
- switch (msg.mMessageType) {
- case SoundPoolMsg::KILL:
- ALOGV("goodbye");
- return NO_ERROR;
- case SoundPoolMsg::LOAD_SAMPLE:
- doLoadSample(msg.mData);
- break;
- default:
- ALOGW("run: Unrecognized message %d\n",
- msg.mMessageType);
- break;
- }
- }
-}
-
-void SoundPoolThread::loadSample(int sampleID) {
- write(SoundPoolMsg(SoundPoolMsg::LOAD_SAMPLE, sampleID));
-}
-
-void SoundPoolThread::doLoadSample(int sampleID) {
- sp <Sample> sample = mSoundPool->findSample(sampleID);
- status_t status = -1;
- if (sample != 0) {
- status = sample->doLoad();
- }
- mSoundPool->notify(SoundPoolEvent(SoundPoolEvent::SAMPLE_LOADED, sampleID, status));
-}
-
-} // end namespace android
diff --git a/media/libmedia/SoundPoolThread.h b/media/libmedia/SoundPoolThread.h
deleted file mode 100644
index 7e96900..0000000
--- a/media/libmedia/SoundPoolThread.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef SOUNDPOOLTHREAD_H_
-#define SOUNDPOOLTHREAD_H_
-
-#include <utils/threads.h>
-#include <utils/Vector.h>
-#include <media/AudioTrack.h>
-
-#include <media/SoundPool.h>
-
-namespace android {
-
-class SoundPoolMsg {
-public:
- enum MessageType { INVALID, KILL, LOAD_SAMPLE };
- SoundPoolMsg() : mMessageType(INVALID), mData(0) {}
- SoundPoolMsg(MessageType MessageType, int data) :
- mMessageType(MessageType), mData(data) {}
- uint16_t mMessageType;
- uint16_t mData;
-};
-
-/*
- * This class handles background requests from the SoundPool
- */
-class SoundPoolThread {
-public:
- SoundPoolThread(SoundPool* SoundPool);
- ~SoundPoolThread();
- void loadSample(int sampleID);
- void quit();
- void write(SoundPoolMsg msg);
-
-private:
- static const size_t maxMessages = 5;
-
- static int beginThread(void* arg);
- int run();
- void doLoadSample(int sampleID);
- const SoundPoolMsg read();
-
- Mutex mLock;
- Condition mCondition;
- Vector<SoundPoolMsg> mMsgQueue;
- SoundPool* mSoundPool;
- bool mRunning;
-};
-
-} // end namespace android
-
-#endif /*SOUNDPOOLTHREAD_H_*/
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 9611ac7..05c89ed 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -835,53 +835,12 @@ void MediaPlayer::notify(int msg, int ext1, int ext2, const Parcel *obj)
}
}
-/*static*/ status_t MediaPlayer::decode(
- const sp<IMediaHTTPService> &httpService,
- const char* url,
- uint32_t *pSampleRate,
- int* pNumChannels,
- audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap,
- size_t *pSize)
-{
- ALOGV("decode(%s)", url);
- status_t status;
- const sp<IMediaPlayerService>& service = getMediaPlayerService();
- if (service != 0) {
- status = service->decode(httpService, url, pSampleRate, pNumChannels, pFormat, heap, pSize);
- } else {
- ALOGE("Unable to locate media service");
- status = DEAD_OBJECT;
- }
- return status;
-
-}
-
void MediaPlayer::died()
{
ALOGV("died");
notify(MEDIA_ERROR, MEDIA_ERROR_SERVER_DIED, 0);
}
-/*static*/ status_t MediaPlayer::decode(int fd, int64_t offset, int64_t length,
- uint32_t *pSampleRate, int* pNumChannels,
- audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap, size_t *pSize)
-{
- ALOGV("decode(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
- status_t status;
- const sp<IMediaPlayerService>& service = getMediaPlayerService();
- if (service != 0) {
- status = service->decode(fd, offset, length, pSampleRate,
- pNumChannels, pFormat, heap, pSize);
- } else {
- ALOGE("Unable to locate media service");
- status = DEAD_OBJECT;
- }
- return status;
-
-}
-
status_t MediaPlayer::setNextMediaPlayer(const sp<MediaPlayer>& next) {
if (mPlayer == NULL) {
return NO_INIT;
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 4d88a7f..1936218 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -384,28 +384,6 @@ sp<IRemoteDisplay> MediaPlayerService::listenForRemoteDisplay(
return new RemoteDisplay(client, iface.string());
}
-status_t MediaPlayerService::AudioCache::dump(int fd, const Vector<String16>& /*args*/) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- result.append(" AudioCache\n");
- if (mHeap != 0) {
- snprintf(buffer, 255, " heap base(%p), size(%zu), flags(%d)\n",
- mHeap->getBase(), mHeap->getSize(), mHeap->getFlags());
- result.append(buffer);
- }
- snprintf(buffer, 255, " msec per frame(%f), channel count(%d), format(%d), frame count(%zd)\n",
- mMsecsPerFrame, mChannelCount, mFormat, mFrameCount);
- result.append(buffer);
- snprintf(buffer, 255, " sample rate(%d), size(%d), error(%d), command complete(%s)\n",
- mSampleRate, mSize, mError, mCommandComplete?"true":"false");
- result.append(buffer);
- ::write(fd, result.string(), result.size());
- return NO_ERROR;
-}
-
status_t MediaPlayerService::AudioOutput::dump(int fd, const Vector<String16>& args) const
{
const size_t SIZE = 256;
@@ -1279,128 +1257,6 @@ int Antagonizer::callbackThread(void* user)
}
#endif
-status_t MediaPlayerService::decode(
- const sp<IMediaHTTPService> &httpService,
- const char* url,
- uint32_t *pSampleRate,
- int* pNumChannels,
- audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap,
- size_t *pSize)
-{
- ALOGV("decode(%s)", url);
- sp<MediaPlayerBase> player;
- status_t status = BAD_VALUE;
-
- // Protect our precious, precious DRMd ringtones by only allowing
- // decoding of http, but not filesystem paths or content Uris.
- // If the application wants to decode those, it should open a
- // filedescriptor for them and use that.
- if (url != NULL && strncmp(url, "http://", 7) != 0) {
- ALOGD("Can't decode %s by path, use filedescriptor instead", url);
- return BAD_VALUE;
- }
-
- player_type playerType =
- MediaPlayerFactory::getPlayerType(NULL /* client */, url);
- ALOGV("player type = %d", playerType);
-
- // create the right type of player
- sp<AudioCache> cache = new AudioCache(heap);
- player = MediaPlayerFactory::createPlayer(playerType, cache.get(), cache->notify);
- if (player == NULL) goto Exit;
- if (player->hardwareOutput()) goto Exit;
-
- static_cast<MediaPlayerInterface*>(player.get())->setAudioSink(cache);
-
- // set data source
- if (player->setDataSource(httpService, url) != NO_ERROR) goto Exit;
-
- ALOGV("prepare");
- player->prepareAsync();
-
- ALOGV("wait for prepare");
- if (cache->wait() != NO_ERROR) goto Exit;
-
- ALOGV("start");
- player->start();
-
- ALOGV("wait for playback complete");
- cache->wait();
- // in case of error, return what was successfully decoded.
- if (cache->size() == 0) {
- goto Exit;
- }
-
- *pSize = cache->size();
- *pSampleRate = cache->sampleRate();
- *pNumChannels = cache->channelCount();
- *pFormat = cache->format();
- ALOGV("return size %d sampleRate=%u, channelCount = %d, format = %d",
- *pSize, *pSampleRate, *pNumChannels, *pFormat);
- status = NO_ERROR;
-
-Exit:
- if (player != 0) player->reset();
- return status;
-}
-
-status_t MediaPlayerService::decode(int fd, int64_t offset, int64_t length,
- uint32_t *pSampleRate, int* pNumChannels,
- audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap, size_t *pSize)
-{
- ALOGV("decode(%d, %lld, %lld)", fd, offset, length);
- sp<MediaPlayerBase> player;
- status_t status = BAD_VALUE;
-
- player_type playerType = MediaPlayerFactory::getPlayerType(NULL /* client */,
- fd,
- offset,
- length);
- ALOGV("player type = %d", playerType);
-
- // create the right type of player
- sp<AudioCache> cache = new AudioCache(heap);
- player = MediaPlayerFactory::createPlayer(playerType, cache.get(), cache->notify);
- if (player == NULL) goto Exit;
- if (player->hardwareOutput()) goto Exit;
-
- static_cast<MediaPlayerInterface*>(player.get())->setAudioSink(cache);
-
- // set data source
- if (player->setDataSource(fd, offset, length) != NO_ERROR) goto Exit;
-
- ALOGV("prepare");
- player->prepareAsync();
-
- ALOGV("wait for prepare");
- if (cache->wait() != NO_ERROR) goto Exit;
-
- ALOGV("start");
- player->start();
-
- ALOGV("wait for playback complete");
- cache->wait();
- // in case of error, return what was successfully decoded.
- if (cache->size() == 0) {
- goto Exit;
- }
-
- *pSize = cache->size();
- *pSampleRate = cache->sampleRate();
- *pNumChannels = cache->channelCount();
- *pFormat = cache->format();
- ALOGV("return size %d, sampleRate=%u, channelCount = %d, format = %d",
- *pSize, *pSampleRate, *pNumChannels, *pFormat);
- status = NO_ERROR;
-
-Exit:
- if (player != 0) player->reset();
- return status;
-}
-
-
#undef LOG_TAG
#define LOG_TAG "AudioSink"
MediaPlayerService::AudioOutput::AudioOutput(int sessionId, int uid, int pid,
@@ -1950,47 +1806,6 @@ uint32_t MediaPlayerService::AudioOutput::getSampleRate() const
return mTrack->getSampleRate();
}
-#undef LOG_TAG
-#define LOG_TAG "AudioCache"
-MediaPlayerService::AudioCache::AudioCache(const sp<IMemoryHeap>& heap) :
- mHeap(heap), mChannelCount(0), mFrameCount(1024), mSampleRate(0), mSize(0),
- mFrameSize(1), mError(NO_ERROR), mCommandComplete(false)
-{
-}
-
-uint32_t MediaPlayerService::AudioCache::latency () const
-{
- return 0;
-}
-
-float MediaPlayerService::AudioCache::msecsPerFrame() const
-{
- return mMsecsPerFrame;
-}
-
-status_t MediaPlayerService::AudioCache::getPosition(uint32_t *position) const
-{
- if (position == 0) return BAD_VALUE;
- *position = mSize / mFrameSize;
- return NO_ERROR;
-}
-
-status_t MediaPlayerService::AudioCache::getTimestamp(AudioTimestamp &ts) const
-{
- ts.mPosition = mSize / mFrameSize;
- nsecs_t now = systemTime(SYSTEM_TIME_MONOTONIC);
- ts.mTime.tv_sec = now / 1000000000LL;
- ts.mTime.tv_nsec = now - (1000000000LL * ts.mTime.tv_sec);
- return NO_ERROR;
-}
-
-status_t MediaPlayerService::AudioCache::getFramesWritten(uint32_t *written) const
-{
- if (written == 0) return BAD_VALUE;
- *written = mSize / mFrameSize;
- return NO_ERROR;
-}
-
////////////////////////////////////////////////////////////////////////////////
struct CallbackThread : public Thread {
@@ -2058,136 +1873,6 @@ bool CallbackThread::threadLoop() {
////////////////////////////////////////////////////////////////////////////////
-status_t MediaPlayerService::AudioCache::open(
- uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
- audio_format_t format, int bufferCount,
- AudioCallback cb, void *cookie, audio_output_flags_t /*flags*/,
- const audio_offload_info_t* /*offloadInfo*/)
-{
- ALOGV("open(%u, %d, 0x%x, %d, %d)", sampleRate, channelCount, channelMask, format, bufferCount);
- if (mHeap->getHeapID() < 0) {
- return NO_INIT;
- }
-
- mSampleRate = sampleRate;
- mChannelCount = (uint16_t)channelCount;
- mFormat = format;
- mMsecsPerFrame = 1.e3 / (float) sampleRate;
- mFrameSize = audio_is_linear_pcm(mFormat)
- ? mChannelCount * audio_bytes_per_sample(mFormat) : 1;
- mFrameCount = mHeap->getSize() / mFrameSize;
-
- if (cb != NULL) {
- mCallbackThread = new CallbackThread(this, cb, cookie);
- }
- return NO_ERROR;
-}
-
-status_t MediaPlayerService::AudioCache::start() {
- if (mCallbackThread != NULL) {
- mCallbackThread->run("AudioCache callback");
- }
- return NO_ERROR;
-}
-
-void MediaPlayerService::AudioCache::stop() {
- if (mCallbackThread != NULL) {
- mCallbackThread->requestExitAndWait();
- }
-}
-
-ssize_t MediaPlayerService::AudioCache::write(const void* buffer, size_t size)
-{
- ALOGV("write(%p, %u)", buffer, size);
- if ((buffer == 0) || (size == 0)) return size;
-
- uint8_t* p = static_cast<uint8_t*>(mHeap->getBase());
- if (p == NULL) return NO_INIT;
- p += mSize;
- ALOGV("memcpy(%p, %p, %u)", p, buffer, size);
-
- bool overflow = mSize + size > mHeap->getSize();
- if (overflow) {
- ALOGE("Heap size overflow! req size: %d, max size: %d", (mSize + size), mHeap->getSize());
- size = mHeap->getSize() - mSize;
- }
- size -= size % mFrameSize; // consume only integral amounts of frame size
- memcpy(p, buffer, size);
- mSize += size;
-
- if (overflow) {
- // Signal heap filled here (last frame may be truncated).
- // After this point, no more data should be written as the
- // heap is filled and the AudioCache should be effectively
- // immutable with respect to future writes.
- //
- // It is thus safe for another thread to read the AudioCache.
- mCommandComplete = true;
- mSignal.signal();
- }
- return size;
-}
-
-// call with lock held
-status_t MediaPlayerService::AudioCache::wait()
-{
- Mutex::Autolock lock(mLock);
- while (!mCommandComplete) {
- mSignal.wait(mLock);
- }
- mCommandComplete = false;
-
- if (mError == NO_ERROR) {
- ALOGV("wait - success");
- } else {
- ALOGV("wait - error");
- }
- return mError;
-}
-
-void MediaPlayerService::AudioCache::notify(
- void* cookie, int msg, int ext1, int ext2, const Parcel* /*obj*/)
-{
- ALOGV("notify(%p, %d, %d, %d)", cookie, msg, ext1, ext2);
- AudioCache* p = static_cast<AudioCache*>(cookie);
-
- // ignore buffering messages
- switch (msg)
- {
- case MEDIA_ERROR:
- ALOGE("Error %d, %d occurred", ext1, ext2);
- p->mError = ext1;
- break;
- case MEDIA_PREPARED:
- ALOGV("prepared");
- break;
- case MEDIA_PLAYBACK_COMPLETE:
- ALOGV("playback complete");
- break;
- default:
- ALOGV("ignored");
- return;
- }
-
- // wake up thread
- Mutex::Autolock lock(p->mLock);
- p->mCommandComplete = true;
- p->mSignal.signal();
-}
-
-int MediaPlayerService::AudioCache::getSessionId() const
-{
- return 0;
-}
-
-uint32_t MediaPlayerService::AudioCache::getSampleRate() const
-{
- if (mMsecsPerFrame == 0) {
- return 0;
- }
- return (uint32_t)(1.e3 / mMsecsPerFrame);
-}
-
void MediaPlayerService::addBatteryData(uint32_t params)
{
Mutex::Autolock lock(mLock);
@@ -2231,7 +1916,7 @@ void MediaPlayerService::addBatteryData(uint32_t params)
return;
}
- // an sudio stream is started
+ // an audio stream is started
if (params & kBatteryDataAudioFlingerStart) {
// record the start time only if currently no other audio
// is being played
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index 3b96e88..fad3447 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -77,7 +77,6 @@ class MediaPlayerService : public BnMediaPlayerService
virtual ~AudioOutput();
virtual bool ready() const { return mTrack != 0; }
- virtual bool realtime() const { return true; }
virtual ssize_t bufferSize() const;
virtual ssize_t frameCount() const;
virtual ssize_t channelCount() const;
@@ -184,75 +183,6 @@ class MediaPlayerService : public BnMediaPlayerService
}; // AudioOutput
- class AudioCache : public MediaPlayerBase::AudioSink
- {
- public:
- AudioCache(const sp<IMemoryHeap>& heap);
- virtual ~AudioCache() {}
-
- virtual bool ready() const { return (mChannelCount > 0) && (mHeap->getHeapID() > 0); }
- virtual bool realtime() const { return false; }
- virtual ssize_t bufferSize() const { return frameSize() * mFrameCount; }
- virtual ssize_t frameCount() const { return mFrameCount; }
- virtual ssize_t channelCount() const { return (ssize_t)mChannelCount; }
- virtual ssize_t frameSize() const { return (ssize_t)mFrameSize; }
- virtual uint32_t latency() const;
- virtual float msecsPerFrame() const;
- virtual status_t getPosition(uint32_t *position) const;
- virtual status_t getTimestamp(AudioTimestamp &ts) const;
- virtual status_t getFramesWritten(uint32_t *frameswritten) const;
- virtual int getSessionId() const;
- virtual uint32_t getSampleRate() const;
-
- virtual status_t open(
- uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
- audio_format_t format, int bufferCount = 1,
- AudioCallback cb = NULL, void *cookie = NULL,
- audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
- const audio_offload_info_t *offloadInfo = NULL);
-
- virtual status_t start();
- virtual ssize_t write(const void* buffer, size_t size);
- virtual void stop();
- virtual void flush() {}
- virtual void pause() {}
- virtual void close() {}
- void setAudioStreamType(audio_stream_type_t streamType __unused) {}
- // stream type is not used for AudioCache
- virtual audio_stream_type_t getAudioStreamType() const { return AUDIO_STREAM_DEFAULT; }
-
- void setVolume(float left __unused, float right __unused) {}
- virtual status_t setPlaybackRatePermille(int32_t ratePermille __unused) { return INVALID_OPERATION; }
- uint32_t sampleRate() const { return mSampleRate; }
- audio_format_t format() const { return mFormat; }
- size_t size() const { return mSize; }
- status_t wait();
-
- sp<IMemoryHeap> getHeap() const { return mHeap; }
-
- static void notify(void* cookie, int msg,
- int ext1, int ext2, const Parcel *obj);
- virtual status_t dump(int fd, const Vector<String16>& args) const;
-
- private:
- AudioCache();
-
- Mutex mLock;
- Condition mSignal;
- sp<IMemoryHeap> mHeap;
- float mMsecsPerFrame;
- uint16_t mChannelCount;
- audio_format_t mFormat;
- ssize_t mFrameCount;
- uint32_t mSampleRate;
- uint32_t mSize;
- size_t mFrameSize;
- int mError;
- bool mCommandComplete;
-
- sp<Thread> mCallbackThread;
- }; // AudioCache
-
public:
static void instantiate();
@@ -263,19 +193,6 @@ public:
virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId);
- virtual status_t decode(
- const sp<IMediaHTTPService> &httpService,
- const char* url,
- uint32_t *pSampleRate,
- int* pNumChannels,
- audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap,
- size_t *pSize);
-
- virtual status_t decode(int fd, int64_t offset, int64_t length,
- uint32_t *pSampleRate, int* pNumChannels,
- audio_format_t* pFormat,
- const sp<IMemoryHeap>& heap, size_t *pSize);
virtual sp<IMediaCodecList> getCodecList() const;
virtual sp<IOMX> getOMX();
virtual sp<ICrypto> makeCrypto();
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index 4bff90f..c52d837 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -637,8 +637,17 @@ void NuPlayerDriver::notifyListener_l(
case MEDIA_PLAYBACK_COMPLETE:
{
if (mState != STATE_RESET_IN_PROGRESS) {
- if (mLooping || (mAutoLoop
- && (mAudioSink == NULL || mAudioSink->realtime()))) {
+ if (mAutoLoop) {
+ audio_stream_type_t streamType = AUDIO_STREAM_MUSIC;
+ if (mAudioSink != NULL) {
+ streamType = mAudioSink->getAudioStreamType();
+ }
+ if (streamType == AUDIO_STREAM_NOTIFICATION) {
+ ALOGW("disabling auto-loop for notification");
+ mAutoLoop = false;
+ }
+ }
+ if (mLooping || mAutoLoop) {
mPlayer->seekToAsync(0);
break;
}
diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index ab8ac79..a31b317 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -879,10 +879,7 @@ void AwesomePlayer::onStreamDone() {
}
if ((mFlags & LOOPING)
- || ((mFlags & AUTO_LOOPING)
- && (mAudioSink == NULL || mAudioSink->realtime()))) {
- // Don't AUTO_LOOP if we're being recorded, since that cannot be
- // turned off and recording would go on indefinitely.
+ || (mFlags & AUTO_LOOPING)) {
seekTo_l(0);