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-rw-r--r--media/libstagefright/NuCachedSource2.cpp34
-rw-r--r--media/libstagefright/codecs/aacdec/SoftAAC2.cpp54
-rw-r--r--media/libstagefright/codecs/aacdec/SoftAAC2.h3
-rw-r--r--media/libstagefright/httplive/LiveSession.cpp63
-rw-r--r--media/libstagefright/httplive/LiveSession.h2
-rw-r--r--media/libstagefright/httplive/PlaylistFetcher.cpp58
-rw-r--r--media/libstagefright/httplive/PlaylistFetcher.h8
-rw-r--r--media/libstagefright/include/NuCachedSource2.h3
8 files changed, 183 insertions, 42 deletions
diff --git a/media/libstagefright/NuCachedSource2.cpp b/media/libstagefright/NuCachedSource2.cpp
index c1feff8..be2a873 100644
--- a/media/libstagefright/NuCachedSource2.cpp
+++ b/media/libstagefright/NuCachedSource2.cpp
@@ -191,6 +191,7 @@ NuCachedSource2::NuCachedSource2(
mFinalStatus(OK),
mLastAccessPos(0),
mFetching(true),
+ mDisconnecting(false),
mLastFetchTimeUs(-1),
mNumRetriesLeft(kMaxNumRetries),
mHighwaterThresholdBytes(kDefaultHighWaterThreshold),
@@ -244,6 +245,23 @@ status_t NuCachedSource2::getEstimatedBandwidthKbps(int32_t *kbps) {
return ERROR_UNSUPPORTED;
}
+void NuCachedSource2::disconnect() {
+ if (mSource->flags() & kIsHTTPBasedSource) {
+ ALOGV("disconnecting HTTPBasedSource");
+
+ {
+ Mutex::Autolock autoLock(mLock);
+ // set mDisconnecting to true, if a fetch returns after
+ // this, the source will be marked as EOS.
+ mDisconnecting = true;
+ }
+
+ // explicitly disconnect from the source, to allow any
+ // pending reads to return more promptly
+ static_cast<HTTPBase *>(mSource.get())->disconnect();
+ }
+}
+
status_t NuCachedSource2::setCacheStatCollectFreq(int32_t freqMs) {
if (mSource->flags() & kIsHTTPBasedSource) {
HTTPBase *source = static_cast<HTTPBase *>(mSource.get());
@@ -327,7 +345,14 @@ void NuCachedSource2::fetchInternal() {
Mutex::Autolock autoLock(mLock);
- if (n < 0) {
+ if (n == 0 || mDisconnecting) {
+ ALOGI("ERROR_END_OF_STREAM");
+
+ mNumRetriesLeft = 0;
+ mFinalStatus = ERROR_END_OF_STREAM;
+
+ mCache->releasePage(page);
+ } else if (n < 0) {
mFinalStatus = n;
if (n == ERROR_UNSUPPORTED || n == -EPIPE) {
// These are errors that are not likely to go away even if we
@@ -337,13 +362,6 @@ void NuCachedSource2::fetchInternal() {
ALOGE("source returned error %zd, %d retries left", n, mNumRetriesLeft);
mCache->releasePage(page);
- } else if (n == 0) {
- ALOGI("ERROR_END_OF_STREAM");
-
- mNumRetriesLeft = 0;
- mFinalStatus = ERROR_END_OF_STREAM;
-
- mCache->releasePage(page);
} else {
if (mFinalStatus != OK) {
ALOGI("retrying a previously failed read succeeded.");
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
index 4569c1c..fb27dca 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -137,6 +137,7 @@ status_t SoftAAC2::initDecoder() {
mOutputDelayRingBuffer = new short[mOutputDelayRingBufferSize];
mOutputDelayRingBufferWritePos = 0;
mOutputDelayRingBufferReadPos = 0;
+ mOutputDelayRingBufferFilled = 0;
if (mAACDecoder == NULL) {
ALOGE("AAC decoder is null. TODO: Can not call aacDecoder_SetParam in the following code");
@@ -408,6 +409,13 @@ void SoftAAC2::configureDownmix() const {
}
bool SoftAAC2::outputDelayRingBufferPutSamples(INT_PCM *samples, int32_t numSamples) {
+ if (numSamples == 0) {
+ return true;
+ }
+ if (outputDelayRingBufferSpaceLeft() < numSamples) {
+ ALOGE("RING BUFFER WOULD OVERFLOW");
+ return false;
+ }
if (mOutputDelayRingBufferWritePos + numSamples <= mOutputDelayRingBufferSize
&& (mOutputDelayRingBufferReadPos <= mOutputDelayRingBufferWritePos
|| mOutputDelayRingBufferReadPos > mOutputDelayRingBufferWritePos + numSamples)) {
@@ -419,10 +427,6 @@ bool SoftAAC2::outputDelayRingBufferPutSamples(INT_PCM *samples, int32_t numSamp
if (mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferSize) {
mOutputDelayRingBufferWritePos -= mOutputDelayRingBufferSize;
}
- if (mOutputDelayRingBufferWritePos == mOutputDelayRingBufferReadPos) {
- ALOGE("RING BUFFER OVERFLOW");
- return false;
- }
} else {
ALOGV("slow SoftAAC2::outputDelayRingBufferPutSamples()");
@@ -432,16 +436,19 @@ bool SoftAAC2::outputDelayRingBufferPutSamples(INT_PCM *samples, int32_t numSamp
if (mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferSize) {
mOutputDelayRingBufferWritePos -= mOutputDelayRingBufferSize;
}
- if (mOutputDelayRingBufferWritePos == mOutputDelayRingBufferReadPos) {
- ALOGE("RING BUFFER OVERFLOW");
- return false;
- }
}
}
+ mOutputDelayRingBufferFilled += numSamples;
return true;
}
int32_t SoftAAC2::outputDelayRingBufferGetSamples(INT_PCM *samples, int32_t numSamples) {
+
+ if (numSamples > mOutputDelayRingBufferFilled) {
+ ALOGE("RING BUFFER WOULD UNDERRUN");
+ return -1;
+ }
+
if (mOutputDelayRingBufferReadPos + numSamples <= mOutputDelayRingBufferSize
&& (mOutputDelayRingBufferWritePos < mOutputDelayRingBufferReadPos
|| mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferReadPos + numSamples)) {
@@ -460,10 +467,6 @@ int32_t SoftAAC2::outputDelayRingBufferGetSamples(INT_PCM *samples, int32_t numS
ALOGV("slow SoftAAC2::outputDelayRingBufferGetSamples()");
for (int32_t i = 0; i < numSamples; i++) {
- if (mOutputDelayRingBufferWritePos == mOutputDelayRingBufferReadPos) {
- ALOGE("RING BUFFER UNDERRUN");
- return -1;
- }
if (samples != 0) {
samples[i] = mOutputDelayRingBuffer[mOutputDelayRingBufferReadPos];
}
@@ -473,22 +476,15 @@ int32_t SoftAAC2::outputDelayRingBufferGetSamples(INT_PCM *samples, int32_t numS
}
}
}
+ mOutputDelayRingBufferFilled -= numSamples;
return numSamples;
}
int32_t SoftAAC2::outputDelayRingBufferSamplesAvailable() {
- int32_t available = mOutputDelayRingBufferWritePos - mOutputDelayRingBufferReadPos;
- if (available < 0) {
- available += mOutputDelayRingBufferSize;
- }
- if (available < 0) {
- ALOGE("FATAL RING BUFFER ERROR");
- return 0;
- }
- return available;
+ return mOutputDelayRingBufferFilled;
}
-int32_t SoftAAC2::outputDelayRingBufferSamplesLeft() {
+int32_t SoftAAC2::outputDelayRingBufferSpaceLeft() {
return mOutputDelayRingBufferSize - outputDelayRingBufferSamplesAvailable();
}
@@ -512,6 +508,11 @@ void SoftAAC2::onQueueFilled(OMX_U32 /* portIndex */) {
OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
mEndOfInput = (inHeader->nFlags & OMX_BUFFERFLAG_EOS) != 0;
+
+ if (mInputBufferCount == 0 && !(inHeader->nFlags & OMX_BUFFERFLAG_CODECCONFIG)) {
+ ALOGE("first buffer should have OMX_BUFFERFLAG_CODECCONFIG set");
+ inHeader->nFlags |= OMX_BUFFERFLAG_CODECCONFIG;
+ }
if ((inHeader->nFlags & OMX_BUFFERFLAG_CODECCONFIG) != 0) {
BufferInfo *inInfo = *inQueue.begin();
OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
@@ -650,7 +651,7 @@ void SoftAAC2::onQueueFilled(OMX_U32 /* portIndex */) {
AAC_DECODER_ERROR decoderErr;
do {
- if (outputDelayRingBufferSamplesLeft() <
+ if (outputDelayRingBufferSpaceLeft() <
(mStreamInfo->frameSize * mStreamInfo->numChannels)) {
ALOGV("skipping decode: not enough space left in ringbuffer");
break;
@@ -705,7 +706,9 @@ void SoftAAC2::onQueueFilled(OMX_U32 /* portIndex */) {
}
// Discard input buffer.
- inHeader->nFilledLen = 0;
+ if (inHeader) {
+ inHeader->nFilledLen = 0;
+ }
aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1);
@@ -736,7 +739,7 @@ void SoftAAC2::onQueueFilled(OMX_U32 /* portIndex */) {
notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
mOutputPortSettingsChange = AWAITING_DISABLED;
- if (inHeader->nFilledLen == 0) {
+ if (inHeader && inHeader->nFilledLen == 0) {
inInfo->mOwnedByUs = false;
mInputBufferCount++;
inQueue.erase(inQueue.begin());
@@ -1022,6 +1025,7 @@ void SoftAAC2::onReset() {
mOutputDelayCompensated = 0;
mOutputDelayRingBufferWritePos = 0;
mOutputDelayRingBufferReadPos = 0;
+ mOutputDelayRingBufferFilled = 0;
mEndOfInput = false;
mEndOfOutput = false;
mBufferTimestamps.clear();
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.h b/media/libstagefright/codecs/aacdec/SoftAAC2.h
index 9fcb598..c3e4459 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.h
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.h
@@ -85,10 +85,11 @@ private:
short *mOutputDelayRingBuffer;
int32_t mOutputDelayRingBufferWritePos;
int32_t mOutputDelayRingBufferReadPos;
+ int32_t mOutputDelayRingBufferFilled;
bool outputDelayRingBufferPutSamples(INT_PCM *samples, int numSamples);
int32_t outputDelayRingBufferGetSamples(INT_PCM *samples, int numSamples);
int32_t outputDelayRingBufferSamplesAvailable();
- int32_t outputDelayRingBufferSamplesLeft();
+ int32_t outputDelayRingBufferSpaceLeft();
DISALLOW_EVIL_CONSTRUCTORS(SoftAAC2);
};
diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index 3720085..a289637 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -63,6 +63,7 @@ LiveSession::LiveSession(
mSwapMask(0),
mCheckBandwidthGeneration(0),
mSwitchGeneration(0),
+ mSubtitleGeneration(0),
mLastDequeuedTimeUs(0ll),
mRealTimeBaseUs(0ll),
mReconfigurationInProgress(false),
@@ -158,9 +159,16 @@ status_t LiveSession::dequeueAccessUnit(
// wait for counterpart
sp<AnotherPacketSource> otherSource;
- if (stream == STREAMTYPE_AUDIO && (mStreamMask & STREAMTYPE_VIDEO)) {
+ uint32_t mask = mNewStreamMask & mStreamMask;
+ uint32_t fetchersMask = 0;
+ for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
+ uint32_t fetcherMask = mFetcherInfos.valueAt(i).mFetcher->getStreamTypeMask();
+ fetchersMask |= fetcherMask;
+ }
+ mask &= fetchersMask;
+ if (stream == STREAMTYPE_AUDIO && (mask & STREAMTYPE_VIDEO)) {
otherSource = mPacketSources.valueFor(STREAMTYPE_VIDEO);
- } else if (stream == STREAMTYPE_VIDEO && (mStreamMask & STREAMTYPE_AUDIO)) {
+ } else if (stream == STREAMTYPE_VIDEO && (mask & STREAMTYPE_AUDIO)) {
otherSource = mPacketSources.valueFor(STREAMTYPE_AUDIO);
}
if (otherSource != NULL && !otherSource->hasBufferAvailable(&finalResult)) {
@@ -282,6 +290,11 @@ status_t LiveSession::dequeueAccessUnit(
mLastDequeuedTimeUs = timeUs;
mRealTimeBaseUs = ALooper::GetNowUs() - timeUs;
} else if (stream == STREAMTYPE_SUBTITLES) {
+ int32_t subtitleGeneration;
+ if ((*accessUnit)->meta()->findInt32("subtitleGeneration", &subtitleGeneration)
+ && subtitleGeneration != mSubtitleGeneration) {
+ return -EAGAIN;
+ };
(*accessUnit)->meta()->setInt32(
"trackIndex", mPlaylist->getSelectedIndex());
(*accessUnit)->meta()->setInt64("baseUs", mRealTimeBaseUs);
@@ -436,6 +449,23 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) {
ALOGE("XXX Received error %d from PlaylistFetcher.", err);
+ // handle EOS on subtitle tracks independently
+ AString uri;
+ if (err == ERROR_END_OF_STREAM && msg->findString("uri", &uri)) {
+ ssize_t i = mFetcherInfos.indexOfKey(uri);
+ if (i >= 0) {
+ const sp<PlaylistFetcher> &fetcher = mFetcherInfos.valueAt(i).mFetcher;
+ if (fetcher != NULL) {
+ uint32_t type = fetcher->getStreamTypeMask();
+ if (type == STREAMTYPE_SUBTITLES) {
+ mPacketSources.valueFor(
+ STREAMTYPE_SUBTITLES)->signalEOS(err);;
+ break;
+ }
+ }
+ }
+ }
+
if (mInPreparationPhase) {
postPrepared(err);
}
@@ -735,7 +765,7 @@ sp<PlaylistFetcher> LiveSession::addFetcher(const char *uri) {
notify->setInt32("switchGeneration", mSwitchGeneration);
FetcherInfo info;
- info.mFetcher = new PlaylistFetcher(notify, this, uri);
+ info.mFetcher = new PlaylistFetcher(notify, this, uri, mSubtitleGeneration);
info.mDurationUs = -1ll;
info.mIsPrepared = false;
info.mToBeRemoved = false;
@@ -1041,6 +1071,24 @@ size_t LiveSession::getBandwidthIndex() {
return index;
}
+int64_t LiveSession::latestMediaSegmentStartTimeUs() {
+ sp<AMessage> audioMeta = mPacketSources.valueFor(STREAMTYPE_AUDIO)->getLatestDequeuedMeta();
+ int64_t minSegmentStartTimeUs = -1, videoSegmentStartTimeUs = -1;
+ if (audioMeta != NULL) {
+ audioMeta->findInt64("segmentStartTimeUs", &minSegmentStartTimeUs);
+ }
+
+ sp<AMessage> videoMeta = mPacketSources.valueFor(STREAMTYPE_VIDEO)->getLatestDequeuedMeta();
+ if (videoMeta != NULL
+ && videoMeta->findInt64("segmentStartTimeUs", &videoSegmentStartTimeUs)) {
+ if (minSegmentStartTimeUs < 0 || videoSegmentStartTimeUs < minSegmentStartTimeUs) {
+ minSegmentStartTimeUs = videoSegmentStartTimeUs;
+ }
+
+ }
+ return minSegmentStartTimeUs;
+}
+
status_t LiveSession::onSeek(const sp<AMessage> &msg) {
int64_t timeUs;
CHECK(msg->findInt64("timeUs", &timeUs));
@@ -1093,6 +1141,11 @@ sp<AMessage> LiveSession::getTrackInfo(size_t trackIndex) const {
}
status_t LiveSession::selectTrack(size_t index, bool select) {
+ if (mPlaylist == NULL) {
+ return INVALID_OPERATION;
+ }
+
+ ++mSubtitleGeneration;
status_t err = mPlaylist->selectTrack(index, select);
if (err == OK) {
sp<AMessage> msg = new AMessage(kWhatChangeConfiguration, id());
@@ -1375,6 +1428,10 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
int32_t discontinuitySeq = -1;
sp<AnotherPacketSource> sources[kMaxStreams];
+ if (i == kSubtitleIndex) {
+ segmentStartTimeUs = latestMediaSegmentStartTimeUs();
+ }
+
// TRICKY: looping from i as earlier streams are already removed from streamMask
for (size_t j = i; j < kMaxStreams; ++j) {
const AString &streamUri = switching ? mStreams[j].mNewUri : mStreams[j].mUri;
diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h
index 6be86cf..7aacca6 100644
--- a/media/libstagefright/httplive/LiveSession.h
+++ b/media/libstagefright/httplive/LiveSession.h
@@ -189,6 +189,7 @@ private:
int32_t mCheckBandwidthGeneration;
int32_t mSwitchGeneration;
+ int32_t mSubtitleGeneration;
size_t mContinuationCounter;
sp<AMessage> mContinuation;
@@ -240,6 +241,7 @@ private:
const char *url, uint8_t *curPlaylistHash, bool *unchanged);
size_t getBandwidthIndex();
+ int64_t latestMediaSegmentStartTimeUs();
static int SortByBandwidth(const BandwidthItem *, const BandwidthItem *);
static StreamType indexToType(int idx);
diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index 1166762..30fa868 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -55,7 +55,8 @@ const int32_t PlaylistFetcher::kNumSkipFrames = 10;
PlaylistFetcher::PlaylistFetcher(
const sp<AMessage> &notify,
const sp<LiveSession> &session,
- const char *uri)
+ const char *uri,
+ int32_t subtitleGeneration)
: mNotify(notify),
mStartTimeUsNotify(notify->dup()),
mSession(session),
@@ -73,6 +74,7 @@ PlaylistFetcher::PlaylistFetcher(
mPrepared(false),
mNextPTSTimeUs(-1ll),
mMonitorQueueGeneration(0),
+ mSubtitleGeneration(subtitleGeneration),
mRefreshState(INITIAL_MINIMUM_RELOAD_DELAY),
mFirstPTSValid(false),
mAbsoluteTimeAnchorUs(0ll),
@@ -1009,7 +1011,16 @@ void PlaylistFetcher::onDownloadNext() {
// bulk extract non-ts files
if (tsBuffer == NULL) {
- err = extractAndQueueAccessUnits(buffer, itemMeta);
+ err = extractAndQueueAccessUnits(buffer, itemMeta);
+ if (err == -EAGAIN) {
+ // starting sequence number too low/high
+ postMonitorQueue();
+ return;
+ } else if (err == ERROR_OUT_OF_RANGE) {
+ // reached stopping point
+ stopAsync(/* clear = */false);
+ return;
+ }
}
if (err != OK) {
@@ -1398,6 +1409,7 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits(
buffer->meta()->setInt64("durationUs", durationUs);
buffer->meta()->setInt64("segmentStartTimeUs", getSegmentStartTimeUs(mSeqNumber));
buffer->meta()->setInt32("discontinuitySeq", mDiscontinuitySeq);
+ buffer->meta()->setInt32("subtitleGeneration", mSubtitleGeneration);
packetSource->queueAccessUnit(buffer);
return OK;
@@ -1552,14 +1564,52 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits(
if (startTimeUs < mStartTimeUs) {
continue;
}
+
+ if (mStartTimeUsNotify != NULL) {
+ int32_t targetDurationSecs;
+ CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs));
+ int64_t targetDurationUs = targetDurationSecs * 1000000ll;
+
+ // Duplicated logic from how we handle .ts playlists.
+ if (mStartup && mSegmentStartTimeUs >= 0
+ && timeUs - mStartTimeUs > targetDurationUs) {
+ int32_t newSeqNumber = getSeqNumberWithAnchorTime(timeUs);
+ if (newSeqNumber >= mSeqNumber) {
+ --mSeqNumber;
+ } else {
+ mSeqNumber = newSeqNumber;
+ }
+ return -EAGAIN;
+ }
+
+ mStartTimeUsNotify->setInt64("timeUsAudio", timeUs);
+ mStartTimeUsNotify->setInt32("discontinuitySeq", mDiscontinuitySeq);
+ mStartTimeUsNotify->setInt32("streamMask", LiveSession::STREAMTYPE_AUDIO);
+ mStartTimeUsNotify->post();
+ mStartTimeUsNotify.clear();
+ }
+ }
+
+ if (mStopParams != NULL) {
+ // Queue discontinuity in original stream.
+ int32_t discontinuitySeq;
+ int64_t stopTimeUs;
+ if (!mStopParams->findInt32("discontinuitySeq", &discontinuitySeq)
+ || discontinuitySeq > mDiscontinuitySeq
+ || !mStopParams->findInt64("timeUsAudio", &stopTimeUs)
+ || (discontinuitySeq == mDiscontinuitySeq && unitTimeUs >= stopTimeUs)) {
+ packetSource->queueAccessUnit(mSession->createFormatChangeBuffer());
+ mStreamTypeMask = 0;
+ mPacketSources.clear();
+ return ERROR_OUT_OF_RANGE;
+ }
}
sp<ABuffer> unit = new ABuffer(aac_frame_length);
memcpy(unit->data(), adtsHeader, aac_frame_length);
unit->meta()->setInt64("timeUs", unitTimeUs);
- unit->meta()->setInt64("segmentStartTimeUs", getSegmentStartTimeUs(mSeqNumber));
- unit->meta()->setInt32("discontinuitySeq", mDiscontinuitySeq);
+ setAccessUnitProperties(unit, packetSource);
packetSource->queueAccessUnit(unit);
}
diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h
index 4ba37fa..78c358f 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.h
+++ b/media/libstagefright/httplive/PlaylistFetcher.h
@@ -49,7 +49,8 @@ struct PlaylistFetcher : public AHandler {
PlaylistFetcher(
const sp<AMessage> &notify,
const sp<LiveSession> &session,
- const char *uri);
+ const char *uri,
+ int32_t subtitleGeneration);
sp<DataSource> getDataSource();
@@ -69,6 +70,10 @@ struct PlaylistFetcher : public AHandler {
void resumeUntilAsync(const sp<AMessage> &params);
+ uint32_t getStreamTypeMask() const {
+ return mStreamTypeMask;
+ }
+
protected:
virtual ~PlaylistFetcher();
virtual void onMessageReceived(const sp<AMessage> &msg);
@@ -129,6 +134,7 @@ private:
int64_t mNextPTSTimeUs;
int32_t mMonitorQueueGeneration;
+ const int32_t mSubtitleGeneration;
enum RefreshState {
INITIAL_MINIMUM_RELOAD_DELAY,
diff --git a/media/libstagefright/include/NuCachedSource2.h b/media/libstagefright/include/NuCachedSource2.h
index 5db4b4b..4252706 100644
--- a/media/libstagefright/include/NuCachedSource2.h
+++ b/media/libstagefright/include/NuCachedSource2.h
@@ -37,6 +37,8 @@ struct NuCachedSource2 : public DataSource {
virtual ssize_t readAt(off64_t offset, void *data, size_t size);
+ virtual void disconnect();
+
virtual status_t getSize(off64_t *size);
virtual uint32_t flags();
@@ -103,6 +105,7 @@ private:
off64_t mLastAccessPos;
sp<AMessage> mAsyncResult;
bool mFetching;
+ bool mDisconnecting;
int64_t mLastFetchTimeUs;
int32_t mNumRetriesLeft;
generated by cgit v1.1 at 2024-07-12 08:00:46 +0000