| /* |
| * 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/stagefright/MediaExtractor.h> |
| #include "SoundPool.h" |
| #include "SoundPoolThread.h" |
| #include <media/AudioPolicyHelper.h> |
| #include <ndk/NdkMediaCodec.h> |
| #include <ndk/NdkMediaExtractor.h> |
| #include <ndk/NdkMediaFormat.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; |
| mMuted = 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; |
| } |
| |
| sp<Sample> SoundPool::findSample(int sampleID) |
| { |
| Mutex::Autolock lock(&mLock); |
| return findSample_l(sampleID); |
| } |
| |
| sp<Sample> SoundPool::findSample_l(int sampleID) |
| { |
| return mSamples.valueFor(sampleID); |
| } |
| |
| 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(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); |
| int sampleID; |
| { |
| Mutex::Autolock lock(&mLock); |
| sampleID = ++mNextSampleID; |
| sp<Sample> sample = new Sample(sampleID, fd, offset, length); |
| mSamples.add(sampleID, sample); |
| sample->startLoad(); |
| } |
| // mDecodeThread->loadSample() must be called outside of mLock. |
| // mDecodeThread->loadSample() may block on mDecodeThread message queue space; |
| // the message queue emptying may block on SoundPool::findSample(). |
| // |
| // It theoretically possible that sample loads might decode out-of-order. |
| mDecodeThread->loadSample(sampleID); |
| return sampleID; |
| } |
| |
| bool SoundPool::unload(int sampleID) |
| { |
| ALOGV("unload: sampleID=%d", sampleID); |
| Mutex::Autolock lock(&mLock); |
| return mSamples.removeItem(sampleID) >= 0; // removeItem() returns index or BAD_VALUE |
| } |
| |
| 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); |
| SoundChannel* channel; |
| int channelID; |
| |
| Mutex::Autolock lock(&mLock); |
| |
| if (mQuit) { |
| return 0; |
| } |
| // is sample ready? |
| sp<Sample> sample(findSample_l(sampleID)); |
| if ((sample == 0) || (sample->state() != Sample::READY)) { |
| ALOGW(" sample %d not READY", sampleID); |
| return 0; |
| } |
| |
| dump(); |
| |
| // allocate a channel |
| channel = allocateChannel_l(priority, sampleID); |
| |
| // 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, int sampleID) |
| { |
| List<SoundChannel*>::iterator iter; |
| SoundChannel* channel = NULL; |
| |
| // check if channel for given sampleID still available |
| if (!mChannels.empty()) { |
| for (iter = mChannels.begin(); iter != mChannels.end(); ++iter) { |
| if (sampleID == (*iter)->getPrevSampleID() && (*iter)->state() == SoundChannel::IDLE) { |
| channel = *iter; |
| mChannels.erase(iter); |
| ALOGV("Allocated recycled channel for same sampleID"); |
| break; |
| } |
| } |
| } |
| |
| // allocate any channel |
| if (!channel && !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::mute(bool muting) |
| { |
| ALOGV("mute(%d)", muting); |
| Mutex::Autolock lock(&mLock); |
| mMuted = muting; |
| if (!mChannels.empty()) { |
| for (List<SoundChannel*>::iterator iter = mChannels.begin(); |
| iter != mChannels.end(); ++iter) { |
| (*iter)->mute(muting); |
| } |
| } |
| } |
| |
| 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, 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; |
| } |
| |
| Sample::~Sample() |
| { |
| ALOGV("Sample::destructor sampleID=%d, fd=%d", mSampleID, mFd); |
| if (mFd > 0) { |
| ALOGV("close(%d)", mFd); |
| ::close(mFd); |
| } |
| } |
| |
| static status_t decode(int fd, int64_t offset, int64_t length, |
| uint32_t *rate, int *numChannels, audio_format_t *audioFormat, |
| sp<MemoryHeapBase> heap, size_t *memsize) { |
| |
| ALOGV("fd %d, offset %" PRId64 ", size %" PRId64, fd, offset, length); |
| AMediaExtractor *ex = AMediaExtractor_new(); |
| status_t err = AMediaExtractor_setDataSourceFd(ex, fd, offset, length); |
| |
| if (err != AMEDIA_OK) { |
| AMediaExtractor_delete(ex); |
| return err; |
| } |
| |
| *audioFormat = AUDIO_FORMAT_PCM_16_BIT; |
| |
| size_t numTracks = AMediaExtractor_getTrackCount(ex); |
| for (size_t i = 0; i < numTracks; i++) { |
| AMediaFormat *format = AMediaExtractor_getTrackFormat(ex, i); |
| const char *mime; |
| if (!AMediaFormat_getString(format, AMEDIAFORMAT_KEY_MIME, &mime)) { |
| AMediaExtractor_delete(ex); |
| AMediaFormat_delete(format); |
| return UNKNOWN_ERROR; |
| } |
| if (strncmp(mime, "audio/", 6) == 0) { |
| |
| AMediaCodec *codec = AMediaCodec_createDecoderByType(mime); |
| if (codec == NULL |
| || AMediaCodec_configure(codec, format, |
| NULL /* window */, NULL /* drm */, 0 /* flags */) != AMEDIA_OK |
| || AMediaCodec_start(codec) != AMEDIA_OK |
| || AMediaExtractor_selectTrack(ex, i) != AMEDIA_OK) { |
| AMediaExtractor_delete(ex); |
| AMediaCodec_delete(codec); |
| AMediaFormat_delete(format); |
| return UNKNOWN_ERROR; |
| } |
| |
| bool sawInputEOS = false; |
| bool sawOutputEOS = false; |
| uint8_t* writePos = static_cast<uint8_t*>(heap->getBase()); |
| size_t available = heap->getSize(); |
| size_t written = 0; |
| |
| AMediaFormat_delete(format); |
| format = AMediaCodec_getOutputFormat(codec); |
| |
| while (!sawOutputEOS) { |
| if (!sawInputEOS) { |
| ssize_t bufidx = AMediaCodec_dequeueInputBuffer(codec, 5000); |
| ALOGV("input buffer %zd", bufidx); |
| if (bufidx >= 0) { |
| size_t bufsize; |
| uint8_t *buf = AMediaCodec_getInputBuffer(codec, bufidx, &bufsize); |
| if (buf == nullptr) { |
| ALOGE("AMediaCodec_getInputBuffer returned nullptr, short decode"); |
| break; |
| } |
| int sampleSize = AMediaExtractor_readSampleData(ex, buf, bufsize); |
| ALOGV("read %d", sampleSize); |
| if (sampleSize < 0) { |
| sampleSize = 0; |
| sawInputEOS = true; |
| ALOGV("EOS"); |
| } |
| int64_t presentationTimeUs = AMediaExtractor_getSampleTime(ex); |
| |
| media_status_t mstatus = AMediaCodec_queueInputBuffer(codec, bufidx, |
| 0 /* offset */, sampleSize, presentationTimeUs, |
| sawInputEOS ? AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM : 0); |
| if (mstatus != AMEDIA_OK) { |
| // AMEDIA_ERROR_UNKNOWN == { -ERANGE -EINVAL -EACCES } |
| ALOGE("AMediaCodec_queueInputBuffer returned status %d, short decode", |
| (int)mstatus); |
| break; |
| } |
| (void)AMediaExtractor_advance(ex); |
| } |
| } |
| |
| AMediaCodecBufferInfo info; |
| int status = AMediaCodec_dequeueOutputBuffer(codec, &info, 1); |
| ALOGV("dequeueoutput returned: %d", status); |
| if (status >= 0) { |
| if (info.flags & AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM) { |
| ALOGV("output EOS"); |
| sawOutputEOS = true; |
| } |
| ALOGV("got decoded buffer size %d", info.size); |
| |
| uint8_t *buf = AMediaCodec_getOutputBuffer(codec, status, NULL /* out_size */); |
| if (buf == nullptr) { |
| ALOGE("AMediaCodec_getOutputBuffer returned nullptr, short decode"); |
| break; |
| } |
| size_t dataSize = info.size; |
| if (dataSize > available) { |
| dataSize = available; |
| } |
| memcpy(writePos, buf + info.offset, dataSize); |
| writePos += dataSize; |
| written += dataSize; |
| available -= dataSize; |
| media_status_t mstatus = AMediaCodec_releaseOutputBuffer( |
| codec, status, false /* render */); |
| if (mstatus != AMEDIA_OK) { |
| // AMEDIA_ERROR_UNKNOWN == { -ERANGE -EINVAL -EACCES } |
| ALOGE("AMediaCodec_releaseOutputBuffer returned status %d, short decode", |
| (int)mstatus); |
| break; |
| } |
| if (available == 0) { |
| // there might be more data, but there's no space for it |
| sawOutputEOS = true; |
| } |
| } else if (status == AMEDIACODEC_INFO_OUTPUT_BUFFERS_CHANGED) { |
| ALOGV("output buffers changed"); |
| } else if (status == AMEDIACODEC_INFO_OUTPUT_FORMAT_CHANGED) { |
| AMediaFormat_delete(format); |
| format = AMediaCodec_getOutputFormat(codec); |
| ALOGV("format changed to: %s", AMediaFormat_toString(format)); |
| } else if (status == AMEDIACODEC_INFO_TRY_AGAIN_LATER) { |
| ALOGV("no output buffer right now"); |
| } else if (status <= AMEDIA_ERROR_BASE) { |
| ALOGE("decode error: %d", status); |
| break; |
| } else { |
| ALOGV("unexpected info code: %d", status); |
| } |
| } |
| |
| (void)AMediaCodec_stop(codec); |
| (void)AMediaCodec_delete(codec); |
| (void)AMediaExtractor_delete(ex); |
| if (!AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_SAMPLE_RATE, (int32_t*) rate) || |
| !AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_CHANNEL_COUNT, numChannels)) { |
| (void)AMediaFormat_delete(format); |
| return UNKNOWN_ERROR; |
| } |
| (void)AMediaFormat_delete(format); |
| *memsize = written; |
| return OK; |
| } |
| (void)AMediaFormat_delete(format); |
| } |
| (void)AMediaExtractor_delete(ex); |
| return UNKNOWN_ERROR; |
| } |
| |
| status_t Sample::doLoad() |
| { |
| uint32_t sampleRate; |
| int numChannels; |
| audio_format_t format; |
| status_t status; |
| mHeap = new MemoryHeapBase(kDefaultHeapSize); |
| |
| ALOGV("Start decode"); |
| status = 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"); |
| 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 > FCC_8)) { |
| 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; |
| mPrevSampleID = -1; |
| } |
| |
| // 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 = NO_ERROR; |
| |
| { // 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); |
| size_t frameCount = 0; |
| |
| if (loop) { |
| const audio_format_t format = sample->format(); |
| const size_t frameSize = audio_is_linear_pcm(format) |
| ? numChannels * audio_bytes_per_sample(format) : 1; |
| frameCount = sample->size() / frameSize; |
| } |
| |
| #ifndef USE_SHARED_MEM_BUFFER |
| uint32_t totalFrames = (kDefaultBufferCount * afFrameCount * sampleRate) / afSampleRate; |
| // Ensure minimum audio buffer size in case of short looped sample |
| if(frameCount < totalFrames) { |
| frameCount = totalFrames; |
| } |
| #endif |
| |
| // check if the existing track has the same sample id. |
| if (mAudioTrack != 0 && mPrevSampleID == sample->sampleID()) { |
| // the sample rate may fail to change if the audio track is a fast track. |
| if (mAudioTrack->setSampleRate(sampleRate) == NO_ERROR) { |
| newTrack = mAudioTrack; |
| ALOGV("reusing track %p for sample %d", mAudioTrack.get(), sample->sampleID()); |
| } |
| } |
| if (newTrack == 0) { |
| // 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, |
| 0 /*default notification frames*/, AUDIO_SESSION_ALLOCATE, |
| AudioTrack::TRANSFER_DEFAULT, |
| NULL /*offloadInfo*/, -1 /*uid*/, -1 /*pid*/, mSoundPool->attributes()); |
| #else |
| uint32_t bufferFrames = (totalFrames + (kDefaultBufferCount - 1)) / kDefaultBufferCount; |
| newTrack = new AudioTrack(streamType, sampleRate, sample->format(), |
| channelMask, frameCount, AUDIO_OUTPUT_FLAG_FAST, callback, userData, |
| bufferFrames, AUDIO_SESSION_ALLOCATE, AudioTrack::TRANSFER_DEFAULT, |
| NULL /*offloadInfo*/, -1 /*uid*/, -1 /*pid*/, mSoundPool->attributes()); |
| #endif |
| oldTrack = mAudioTrack; |
| status = newTrack->initCheck(); |
| if (status != NO_ERROR) { |
| ALOGE("Error creating AudioTrack"); |
| // newTrack goes out of scope, so reference count drops to zero |
| goto exit; |
| } |
| // From now on, AudioTrack callbacks received with previous toggle value will be ignored. |
| mToggle = toggle; |
| mAudioTrack = newTrack; |
| ALOGV("using new track %p for sample %d", newTrack.get(), sample->sampleID()); |
| } |
| newTrack->setVolume(leftVolume, rightVolume); |
| newTrack->setLoop(0, frameCount, loop); |
| 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) { |
| ALOGV("process %p channel %d event %s", |
| this, mChannelID, (event == AudioTrack::EVENT_UNDERRUN) ? "UNDERRUN" : |
| "BUFFER_END"); |
| mSoundPool->addToStopList(this); |
| } else if (event == AudioTrack::EVENT_LOOP_END) { |
| ALOGV("End loop %p channel %d", this, mChannelID); |
| } else if (event == AudioTrack::EVENT_NEW_IAUDIOTRACK) { |
| ALOGV("process %p channel %d NEW_IAUDIOTRACK", 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(); |
| mPrevSampleID = mSample->sampleID(); |
| 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 && !mMuted) |
| mAudioTrack->setVolume(leftVolume, rightVolume); |
| } |
| |
| void SoundChannel::setVolume(float leftVolume, float rightVolume) |
| { |
| Mutex::Autolock lock(&mLock); |
| setVolume_l(leftVolume, rightVolume); |
| } |
| |
| void SoundChannel::mute(bool muting) |
| { |
| Mutex::Autolock lock(&mLock); |
| mMuted = muting; |
| if (mAudioTrack != NULL) { |
| if (mMuted) { |
| mAudioTrack->setVolume(0.0f, 0.0f); |
| } else { |
| mAudioTrack->setVolume(mLeftVolume, mRightVolume); |
| } |
| } |
| } |
| |
| 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 |