auto import from //depot/cupcake/@135843
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
new file mode 100644
index 0000000..e79f336
--- /dev/null
+++ b/media/libmedia/AudioTrack.cpp
@@ -0,0 +1,1021 @@
+/* //device/extlibs/pv/android/AudioTrack.cpp
+**
+** Copyright 2007, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+** http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AudioTrack"
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <limits.h>
+
+#include <sched.h>
+#include <sys/resource.h>
+
+#include <private/media/AudioTrackShared.h>
+
+#include <media/AudioSystem.h>
+#include <media/AudioTrack.h>
+
+#include <utils/Log.h>
+#include <utils/MemoryDealer.h>
+#include <utils/Parcel.h>
+#include <utils/IPCThreadState.h>
+#include <utils/Timers.h>
+#include <cutils/atomic.h>
+
+#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
+#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+
+namespace android {
+
+// ---------------------------------------------------------------------------
+
+AudioTrack::AudioTrack()
+ : mStatus(NO_INIT)
+{
+}
+
+AudioTrack::AudioTrack(
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount,
+ uint32_t flags,
+ callback_t cbf,
+ void* user,
+ int notificationFrames)
+ : mStatus(NO_INIT)
+{
+ mStatus = set(streamType, sampleRate, format, channelCount,
+ frameCount, flags, cbf, user, notificationFrames, 0);
+}
+
+AudioTrack::AudioTrack(
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ const sp<IMemory>& sharedBuffer,
+ uint32_t flags,
+ callback_t cbf,
+ void* user,
+ int notificationFrames)
+ : mStatus(NO_INIT)
+{
+ mStatus = set(streamType, sampleRate, format, channelCount,
+ 0, flags, cbf, user, notificationFrames, sharedBuffer);
+}
+
+AudioTrack::~AudioTrack()
+{
+ LOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer());
+
+ if (mStatus == NO_ERROR) {
+ // Make sure that callback function exits in the case where
+ // it is looping on buffer full condition in obtainBuffer().
+ // Otherwise the callback thread will never exit.
+ stop();
+ if (mAudioTrackThread != 0) {
+ mCblk->cv.signal();
+ mAudioTrackThread->requestExitAndWait();
+ mAudioTrackThread.clear();
+ }
+ mAudioTrack.clear();
+ IPCThreadState::self()->flushCommands();
+ }
+}
+
+status_t AudioTrack::set(
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount,
+ uint32_t flags,
+ callback_t cbf,
+ void* user,
+ int notificationFrames,
+ const sp<IMemory>& sharedBuffer,
+ bool threadCanCallJava)
+{
+
+ LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
+
+ if (mAudioFlinger != 0) {
+ LOGE("Track already in use");
+ return INVALID_OPERATION;
+ }
+
+ const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
+ if (audioFlinger == 0) {
+ LOGE("Could not get audioflinger");
+ return NO_INIT;
+ }
+ int afSampleRate;
+ if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
+ return NO_INIT;
+ }
+ int afFrameCount;
+ if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
+ return NO_INIT;
+ }
+ uint32_t afLatency;
+ if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
+ return NO_INIT;
+ }
+
+ // handle default values first.
+ if (streamType == AudioSystem::DEFAULT) {
+ streamType = AudioSystem::MUSIC;
+ }
+ if (sampleRate == 0) {
+ sampleRate = afSampleRate;
+ }
+ // these below should probably come from the audioFlinger too...
+ if (format == 0) {
+ format = AudioSystem::PCM_16_BIT;
+ }
+ if (channelCount == 0) {
+ channelCount = 2;
+ }
+
+ // validate parameters
+ if (((format != AudioSystem::PCM_8_BIT) || sharedBuffer != 0) &&
+ (format != AudioSystem::PCM_16_BIT)) {
+ LOGE("Invalid format");
+ return BAD_VALUE;
+ }
+ if (channelCount != 1 && channelCount != 2) {
+ LOGE("Invalid channel number");
+ return BAD_VALUE;
+ }
+
+ // Ensure that buffer depth covers at least audio hardware latency
+ uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
+ if (minBufCount < 2) minBufCount = 2;
+
+ // When playing from shared buffer, playback will start even if last audioflinger
+ // block is partly filled.
+ if (sharedBuffer != 0 && minBufCount > 1) {
+ minBufCount--;
+ }
+
+ int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
+
+ if (sharedBuffer == 0) {
+ if (frameCount == 0) {
+ frameCount = minFrameCount;
+ }
+ if (notificationFrames == 0) {
+ notificationFrames = frameCount/2;
+ }
+ // Make sure that application is notified with sufficient margin
+ // before underrun
+ if (notificationFrames > frameCount/2) {
+ notificationFrames = frameCount/2;
+ }
+ } else {
+ // Ensure that buffer alignment matches channelcount
+ if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
+ LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
+ return BAD_VALUE;
+ }
+ frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
+ }
+
+ if (frameCount < minFrameCount) {
+ LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
+ return BAD_VALUE;
+ }
+
+ // create the track
+ status_t status;
+ sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
+ streamType, sampleRate, format, channelCount, frameCount, flags, sharedBuffer, &status);
+
+ if (track == 0) {
+ LOGE("AudioFlinger could not create track, status: %d", status);
+ return status;
+ }
+ sp<IMemory> cblk = track->getCblk();
+ if (cblk == 0) {
+ LOGE("Could not get control block");
+ return NO_INIT;
+ }
+ if (cbf != 0) {
+ mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
+ if (mAudioTrackThread == 0) {
+ LOGE("Could not create callback thread");
+ return NO_INIT;
+ }
+ }
+
+ mStatus = NO_ERROR;
+
+ mAudioFlinger = audioFlinger;
+ mAudioTrack = track;
+ mCblkMemory = cblk;
+ mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
+ mCblk->out = 1;
+ // Update buffer size in case it has been limited by AudioFlinger during track creation
+ mFrameCount = mCblk->frameCount;
+ if (sharedBuffer == 0) {
+ mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+ } else {
+ mCblk->buffers = sharedBuffer->pointer();
+ // Force buffer full condition as data is already present in shared memory
+ mCblk->stepUser(mFrameCount);
+ }
+ mCblk->volume[0] = mCblk->volume[1] = 0x1000;
+ mVolume[LEFT] = 1.0f;
+ mVolume[RIGHT] = 1.0f;
+ mSampleRate = sampleRate;
+ mStreamType = streamType;
+ mFormat = format;
+ mChannelCount = channelCount;
+ mSharedBuffer = sharedBuffer;
+ mMuted = false;
+ mActive = 0;
+ mCbf = cbf;
+ mNotificationFrames = notificationFrames;
+ mRemainingFrames = notificationFrames;
+ mUserData = user;
+ mLatency = afLatency + (1000*mFrameCount) / mSampleRate;
+ mLoopCount = 0;
+ mMarkerPosition = 0;
+ mNewPosition = 0;
+ mUpdatePeriod = 0;
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::initCheck() const
+{
+ return mStatus;
+}
+
+// -------------------------------------------------------------------------
+
+uint32_t AudioTrack::latency() const
+{
+ return mLatency;
+}
+
+int AudioTrack::streamType() const
+{
+ return mStreamType;
+}
+
+uint32_t AudioTrack::sampleRate() const
+{
+ return mSampleRate;
+}
+
+int AudioTrack::format() const
+{
+ return mFormat;
+}
+
+int AudioTrack::channelCount() const
+{
+ return mChannelCount;
+}
+
+uint32_t AudioTrack::frameCount() const
+{
+ return mFrameCount;
+}
+
+int AudioTrack::frameSize() const
+{
+ return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
+}
+
+sp<IMemory>& AudioTrack::sharedBuffer()
+{
+ return mSharedBuffer;
+}
+
+// -------------------------------------------------------------------------
+
+void AudioTrack::start()
+{
+ sp<AudioTrackThread> t = mAudioTrackThread;
+
+ LOGV("start %p", this);
+ if (t != 0) {
+ if (t->exitPending()) {
+ if (t->requestExitAndWait() == WOULD_BLOCK) {
+ LOGE("AudioTrack::start called from thread");
+ return;
+ }
+ }
+ t->mLock.lock();
+ }
+
+ if (android_atomic_or(1, &mActive) == 0) {
+ mNewPosition = mCblk->server + mUpdatePeriod;
+ mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
+ mCblk->waitTimeMs = 0;
+ if (t != 0) {
+ t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT);
+ } else {
+ setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
+ }
+ mAudioTrack->start();
+ }
+
+ if (t != 0) {
+ t->mLock.unlock();
+ }
+}
+
+void AudioTrack::stop()
+{
+ sp<AudioTrackThread> t = mAudioTrackThread;
+
+ LOGV("stop %p", this);
+ if (t != 0) {
+ t->mLock.lock();
+ }
+
+ if (android_atomic_and(~1, &mActive) == 1) {
+ mAudioTrack->stop();
+ // Cancel loops (If we are in the middle of a loop, playback
+ // would not stop until loopCount reaches 0).
+ setLoop(0, 0, 0);
+ // Force flush if a shared buffer is used otherwise audioflinger
+ // will not stop before end of buffer is reached.
+ if (mSharedBuffer != 0) {
+ flush();
+ }
+ if (t != 0) {
+ t->requestExit();
+ } else {
+ setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
+ }
+ }
+
+ if (t != 0) {
+ t->mLock.unlock();
+ }
+}
+
+bool AudioTrack::stopped() const
+{
+ return !mActive;
+}
+
+void AudioTrack::flush()
+{
+ LOGV("flush");
+
+ if (!mActive) {
+ mCblk->lock.lock();
+ mAudioTrack->flush();
+ // Release AudioTrack callback thread in case it was waiting for new buffers
+ // in AudioTrack::obtainBuffer()
+ mCblk->cv.signal();
+ mCblk->lock.unlock();
+ }
+}
+
+void AudioTrack::pause()
+{
+ LOGV("pause");
+ if (android_atomic_and(~1, &mActive) == 1) {
+ mActive = 0;
+ mAudioTrack->pause();
+ }
+}
+
+void AudioTrack::mute(bool e)
+{
+ mAudioTrack->mute(e);
+ mMuted = e;
+}
+
+bool AudioTrack::muted() const
+{
+ return mMuted;
+}
+
+void AudioTrack::setVolume(float left, float right)
+{
+ mVolume[LEFT] = left;
+ mVolume[RIGHT] = right;
+
+ // write must be atomic
+ mCblk->volumeLR = (int32_t(int16_t(left * 0x1000)) << 16) | int16_t(right * 0x1000);
+}
+
+void AudioTrack::getVolume(float* left, float* right)
+{
+ *left = mVolume[LEFT];
+ *right = mVolume[RIGHT];
+}
+
+void AudioTrack::setSampleRate(int rate)
+{
+ int afSamplingRate;
+
+ if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
+ return;
+ }
+ // Resampler implementation limits input sampling rate to 2 x output sampling rate.
+ if (rate <= 0) rate = 1;
+ if (rate > afSamplingRate*2) rate = afSamplingRate*2;
+ if (rate > MAX_SAMPLE_RATE) rate = MAX_SAMPLE_RATE;
+
+ mCblk->sampleRate = rate;
+}
+
+uint32_t AudioTrack::getSampleRate()
+{
+ return uint32_t(mCblk->sampleRate);
+}
+
+status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
+{
+ audio_track_cblk_t* cblk = mCblk;
+
+
+ Mutex::Autolock _l(cblk->lock);
+
+ if (loopCount == 0) {
+ cblk->loopStart = UINT_MAX;
+ cblk->loopEnd = UINT_MAX;
+ cblk->loopCount = 0;
+ mLoopCount = 0;
+ return NO_ERROR;
+ }
+
+ if (loopStart >= loopEnd ||
+ loopEnd - loopStart > mFrameCount) {
+ LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, mFrameCount, cblk->user);
+ return BAD_VALUE;
+ }
+
+ if ((mSharedBuffer != 0) && (loopEnd > mFrameCount)) {
+ LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
+ loopStart, loopEnd, mFrameCount);
+ return BAD_VALUE;
+ }
+
+ cblk->loopStart = loopStart;
+ cblk->loopEnd = loopEnd;
+ cblk->loopCount = loopCount;
+ mLoopCount = loopCount;
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount)
+{
+ if (loopStart != 0) {
+ *loopStart = mCblk->loopStart;
+ }
+ if (loopEnd != 0) {
+ *loopEnd = mCblk->loopEnd;
+ }
+ if (loopCount != 0) {
+ if (mCblk->loopCount < 0) {
+ *loopCount = -1;
+ } else {
+ *loopCount = mCblk->loopCount;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::setMarkerPosition(uint32_t marker)
+{
+ if (mCbf == 0) return INVALID_OPERATION;
+
+ mMarkerPosition = marker;
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::getMarkerPosition(uint32_t *marker)
+{
+ if (marker == 0) return BAD_VALUE;
+
+ *marker = mMarkerPosition;
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
+{
+ if (mCbf == 0) return INVALID_OPERATION;
+
+ uint32_t curPosition;
+ getPosition(&curPosition);
+ mNewPosition = curPosition + updatePeriod;
+ mUpdatePeriod = updatePeriod;
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod)
+{
+ if (updatePeriod == 0) return BAD_VALUE;
+
+ *updatePeriod = mUpdatePeriod;
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::setPosition(uint32_t position)
+{
+ Mutex::Autolock _l(mCblk->lock);
+
+ if (!stopped()) return INVALID_OPERATION;
+
+ if (position > mCblk->user) return BAD_VALUE;
+
+ mCblk->server = position;
+ mCblk->forceReady = 1;
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::getPosition(uint32_t *position)
+{
+ if (position == 0) return BAD_VALUE;
+
+ *position = mCblk->server;
+
+ return NO_ERROR;
+}
+
+status_t AudioTrack::reload()
+{
+ if (!stopped()) return INVALID_OPERATION;
+
+ flush();
+
+ mCblk->stepUser(mFrameCount);
+
+ return NO_ERROR;
+}
+
+// -------------------------------------------------------------------------
+
+status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
+{
+ int active;
+ int timeout = 0;
+ status_t result;
+ audio_track_cblk_t* cblk = mCblk;
+ uint32_t framesReq = audioBuffer->frameCount;
+
+ audioBuffer->frameCount = 0;
+ audioBuffer->size = 0;
+
+ uint32_t framesAvail = cblk->framesAvailable();
+
+ if (framesAvail == 0) {
+ Mutex::Autolock _l(cblk->lock);
+ goto start_loop_here;
+ while (framesAvail == 0) {
+ active = mActive;
+ if (UNLIKELY(!active)) {
+ LOGV("Not active and NO_MORE_BUFFERS");
+ return NO_MORE_BUFFERS;
+ }
+ if (UNLIKELY(!waitCount))
+ return WOULD_BLOCK;
+ timeout = 0;
+ result = cblk->cv.waitRelative(cblk->lock, milliseconds(WAIT_PERIOD_MS));
+ if (__builtin_expect(result!=NO_ERROR, false)) {
+ cblk->waitTimeMs += WAIT_PERIOD_MS;
+ if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
+ // timing out when a loop has been set and we have already written upto loop end
+ // is a normal condition: no need to wake AudioFlinger up.
+ if (cblk->user < cblk->loopEnd) {
+ LOGW( "obtainBuffer timed out (is the CPU pegged?) %p "
+ "user=%08x, server=%08x", this, cblk->user, cblk->server);
+ //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
+ cblk->lock.unlock();
+ mAudioTrack->start();
+ cblk->lock.lock();
+ timeout = 1;
+ }
+ cblk->waitTimeMs = 0;
+ }
+
+ if (--waitCount == 0) {
+ return TIMED_OUT;
+ }
+ }
+ // read the server count again
+ start_loop_here:
+ framesAvail = cblk->framesAvailable_l();
+ }
+ }
+
+ cblk->waitTimeMs = 0;
+
+ if (framesReq > framesAvail) {
+ framesReq = framesAvail;
+ }
+
+ uint32_t u = cblk->user;
+ uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
+
+ if (u + framesReq > bufferEnd) {
+ framesReq = bufferEnd - u;
+ }
+
+ LOGW_IF(timeout,
+ "*** SERIOUS WARNING *** obtainBuffer() timed out "
+ "but didn't need to be locked. We recovered, but "
+ "this shouldn't happen (user=%08x, server=%08x)", cblk->user, cblk->server);
+
+ audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
+ audioBuffer->channelCount= mChannelCount;
+ audioBuffer->format = AudioSystem::PCM_16_BIT;
+ audioBuffer->frameCount = framesReq;
+ audioBuffer->size = framesReq*mChannelCount*sizeof(int16_t);
+ audioBuffer->raw = (int8_t *)cblk->buffer(u);
+ active = mActive;
+ return active ? status_t(NO_ERROR) : status_t(STOPPED);
+}
+
+void AudioTrack::releaseBuffer(Buffer* audioBuffer)
+{
+ audio_track_cblk_t* cblk = mCblk;
+ cblk->stepUser(audioBuffer->frameCount);
+}
+
+// -------------------------------------------------------------------------
+
+ssize_t AudioTrack::write(const void* buffer, size_t userSize)
+{
+
+ if (mSharedBuffer != 0) return INVALID_OPERATION;
+
+ if (ssize_t(userSize) < 0) {
+ // sanity-check. user is most-likely passing an error code.
+ LOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
+ buffer, userSize, userSize);
+ return BAD_VALUE;
+ }
+
+ LOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive);
+
+ ssize_t written = 0;
+ const int8_t *src = (const int8_t *)buffer;
+ Buffer audioBuffer;
+
+ do {
+ audioBuffer.frameCount = userSize/mChannelCount;
+ if (mFormat == AudioSystem::PCM_16_BIT) {
+ audioBuffer.frameCount >>= 1;
+ }
+ // Calling obtainBuffer() with a negative wait count causes
+ // an (almost) infinite wait time.
+ status_t err = obtainBuffer(&audioBuffer, -1);
+ if (err < 0) {
+ // out of buffers, return #bytes written
+ if (err == status_t(NO_MORE_BUFFERS))
+ break;
+ return ssize_t(err);
+ }
+
+ size_t toWrite;
+ if (mFormat == AudioSystem::PCM_8_BIT) {
+ // Divide capacity by 2 to take expansion into account
+ toWrite = audioBuffer.size>>1;
+ // 8 to 16 bit conversion
+ int count = toWrite;
+ int16_t *dst = (int16_t *)(audioBuffer.i8);
+ while(count--) {
+ *dst++ = (int16_t)(*src++^0x80) << 8;
+ }
+ }else {
+ toWrite = audioBuffer.size;
+ memcpy(audioBuffer.i8, src, toWrite);
+ src += toWrite;
+ }
+ userSize -= toWrite;
+ written += toWrite;
+
+ releaseBuffer(&audioBuffer);
+ } while (userSize);
+
+ return written;
+}
+
+// -------------------------------------------------------------------------
+
+bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
+{
+ Buffer audioBuffer;
+ uint32_t frames;
+ size_t writtenSize;
+
+ // Manage underrun callback
+ if (mActive && (mCblk->framesReady() == 0)) {
+ LOGV("Underrun user: %x, server: %x, flowControlFlag %d", mCblk->user, mCblk->server, mCblk->flowControlFlag);
+ if (mCblk->flowControlFlag == 0) {
+ mCbf(EVENT_UNDERRUN, mUserData, 0);
+ if (mCblk->server == mCblk->frameCount) {
+ mCbf(EVENT_BUFFER_END, mUserData, 0);
+ }
+ mCblk->flowControlFlag = 1;
+ if (mSharedBuffer != 0) return false;
+ }
+ }
+
+ // Manage loop end callback
+ while (mLoopCount > mCblk->loopCount) {
+ int loopCount = -1;
+ mLoopCount--;
+ if (mLoopCount >= 0) loopCount = mLoopCount;
+
+ mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
+ }
+
+ // Manage marker callback
+ if(mMarkerPosition > 0) {
+ if (mCblk->server >= mMarkerPosition) {
+ mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
+ mMarkerPosition = 0;
+ }
+ }
+
+ // Manage new position callback
+ if(mUpdatePeriod > 0) {
+ while (mCblk->server >= mNewPosition) {
+ mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
+ mNewPosition += mUpdatePeriod;
+ }
+ }
+
+ // If Shared buffer is used, no data is requested from client.
+ if (mSharedBuffer != 0) {
+ frames = 0;
+ } else {
+ frames = mRemainingFrames;
+ }
+
+ do {
+
+ audioBuffer.frameCount = frames;
+
+ // Calling obtainBuffer() with a wait count of 1
+ // limits wait time to WAIT_PERIOD_MS. This prevents from being
+ // stuck here not being able to handle timed events (position, markers, loops).
+ status_t err = obtainBuffer(&audioBuffer, 1);
+ if (err < NO_ERROR) {
+ if (err != TIMED_OUT) {
+ LOGE("Error obtaining an audio buffer, giving up.");
+ return false;
+ }
+ break;
+ }
+ if (err == status_t(STOPPED)) return false;
+
+ // Divide buffer size by 2 to take into account the expansion
+ // due to 8 to 16 bit conversion: the callback must fill only half
+ // of the destination buffer
+ if (mFormat == AudioSystem::PCM_8_BIT) {
+ audioBuffer.size >>= 1;
+ }
+
+ size_t reqSize = audioBuffer.size;
+ mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
+ writtenSize = audioBuffer.size;
+
+ // Sanity check on returned size
+ if (ssize_t(writtenSize) <= 0) break;
+ if (writtenSize > reqSize) writtenSize = reqSize;
+
+ if (mFormat == AudioSystem::PCM_8_BIT) {
+ // 8 to 16 bit conversion
+ const int8_t *src = audioBuffer.i8 + writtenSize-1;
+ int count = writtenSize;
+ int16_t *dst = audioBuffer.i16 + writtenSize-1;
+ while(count--) {
+ *dst-- = (int16_t)(*src--^0x80) << 8;
+ }
+ writtenSize <<= 1;
+ }
+
+ audioBuffer.size = writtenSize;
+ audioBuffer.frameCount = writtenSize/mChannelCount/sizeof(int16_t);
+ frames -= audioBuffer.frameCount;
+
+ releaseBuffer(&audioBuffer);
+ }
+ while (frames);
+
+ if (frames == 0) {
+ mRemainingFrames = mNotificationFrames;
+ } else {
+ mRemainingFrames = frames;
+ }
+ return true;
+}
+
+status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
+{
+
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ result.append(" AudioTrack::dump\n");
+ snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]);
+ result.append(buffer);
+ snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mFrameCount);
+ result.append(buffer);
+ snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", mSampleRate, mStatus, mMuted);
+ result.append(buffer);
+ snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency);
+ result.append(buffer);
+ ::write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+// =========================================================================
+
+AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
+ : Thread(bCanCallJava), mReceiver(receiver)
+{
+}
+
+bool AudioTrack::AudioTrackThread::threadLoop()
+{
+ return mReceiver.processAudioBuffer(this);
+}
+
+status_t AudioTrack::AudioTrackThread::readyToRun()
+{
+ return NO_ERROR;
+}
+
+void AudioTrack::AudioTrackThread::onFirstRef()
+{
+}
+
+// =========================================================================
+
+audio_track_cblk_t::audio_track_cblk_t()
+ : user(0), server(0), userBase(0), serverBase(0), buffers(0), frameCount(0),
+ loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0), flowControlFlag(1), forceReady(0)
+{
+}
+
+uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
+{
+ uint32_t u = this->user;
+
+ u += frameCount;
+ // Ensure that user is never ahead of server for AudioRecord
+ if (out) {
+ // If stepServer() has been called once, switch to normal obtainBuffer() timeout period
+ if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
+ bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
+ }
+ } else if (u > this->server) {
+ LOGW("stepServer occured after track reset");
+ u = this->server;
+ }
+
+ if (u >= userBase + this->frameCount) {
+ userBase += this->frameCount;
+ }
+
+ this->user = u;
+
+ // Clear flow control error condition as new data has been written/read to/from buffer.
+ flowControlFlag = 0;
+
+ return u;
+}
+
+bool audio_track_cblk_t::stepServer(uint32_t frameCount)
+{
+ // the code below simulates lock-with-timeout
+ // we MUST do this to protect the AudioFlinger server
+ // as this lock is shared with the client.
+ status_t err;
+
+ err = lock.tryLock();
+ if (err == -EBUSY) { // just wait a bit
+ usleep(1000);
+ err = lock.tryLock();
+ }
+ if (err != NO_ERROR) {
+ // probably, the client just died.
+ return false;
+ }
+
+ uint32_t s = this->server;
+
+ s += frameCount;
+ if (out) {
+ // Mark that we have read the first buffer so that next time stepUser() is called
+ // we switch to normal obtainBuffer() timeout period
+ if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
+ bufferTimeoutMs = MAX_RUN_TIMEOUT_MS - 1;
+ }
+ // It is possible that we receive a flush()
+ // while the mixer is processing a block: in this case,
+ // stepServer() is called After the flush() has reset u & s and
+ // we have s > u
+ if (s > this->user) {
+ LOGW("stepServer occured after track reset");
+ s = this->user;
+ }
+ }
+
+ if (s >= loopEnd) {
+ LOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
+ s = loopStart;
+ if (--loopCount == 0) {
+ loopEnd = UINT_MAX;
+ loopStart = UINT_MAX;
+ }
+ }
+ if (s >= serverBase + this->frameCount) {
+ serverBase += this->frameCount;
+ }
+
+ this->server = s;
+
+ cv.signal();
+ lock.unlock();
+ return true;
+}
+
+void* audio_track_cblk_t::buffer(uint32_t offset) const
+{
+ return (int16_t *)this->buffers + (offset-userBase)*this->channels;
+}
+
+uint32_t audio_track_cblk_t::framesAvailable()
+{
+ Mutex::Autolock _l(lock);
+ return framesAvailable_l();
+}
+
+uint32_t audio_track_cblk_t::framesAvailable_l()
+{
+ uint32_t u = this->user;
+ uint32_t s = this->server;
+
+ if (out) {
+ uint32_t limit = (s < loopStart) ? s : loopStart;
+ return limit + frameCount - u;
+ } else {
+ return frameCount + u - s;
+ }
+}
+
+uint32_t audio_track_cblk_t::framesReady()
+{
+ uint32_t u = this->user;
+ uint32_t s = this->server;
+
+ if (out) {
+ if (u < loopEnd) {
+ return u - s;
+ } else {
+ Mutex::Autolock _l(lock);
+ if (loopCount >= 0) {
+ return (loopEnd - loopStart)*loopCount + u - s;
+ } else {
+ return UINT_MAX;
+ }
+ }
+ } else {
+ return s - u;
+ }
+}
+
+// -------------------------------------------------------------------------
+
+}; // namespace android
+