Merge "HLS: bandwidth estimator changes"
diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index a94754b..f5328a6 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -50,13 +50,75 @@
namespace android {
// static
-// Number of recently-read bytes to use for bandwidth estimation
-const size_t LiveSession::kBandwidthHistoryBytes = 200 * 1024;
// High water mark to start up switch or report prepared)
const int64_t LiveSession::kHighWaterMark = 8000000ll;
const int64_t LiveSession::kMidWaterMark = 5000000ll;
const int64_t LiveSession::kLowWaterMark = 3000000ll;
+struct LiveSession::BandwidthEstimator : public RefBase {
+ BandwidthEstimator();
+
+ void addBandwidthMeasurement(size_t numBytes, int64_t delayUs);
+ bool estimateBandwidth(int32_t *bandwidth);
+
+private:
+ // Bandwidth estimation parameters
+ static const int32_t kMaxBandwidthHistoryItems = 20;
+ static const int64_t kMaxBandwidthHistoryWindowUs = 3000000ll; // 3 sec
+
+ struct BandwidthEntry {
+ int64_t mDelayUs;
+ size_t mNumBytes;
+ };
+
+ Mutex mLock;
+ List<BandwidthEntry> mBandwidthHistory;
+ int64_t mTotalTransferTimeUs;
+ size_t mTotalTransferBytes;
+
+ DISALLOW_EVIL_CONSTRUCTORS(BandwidthEstimator);
+};
+
+LiveSession::BandwidthEstimator::BandwidthEstimator() :
+ mTotalTransferTimeUs(0),
+ mTotalTransferBytes(0) {
+}
+
+void LiveSession::BandwidthEstimator::addBandwidthMeasurement(
+ size_t numBytes, int64_t delayUs) {
+ AutoMutex autoLock(mLock);
+
+ BandwidthEntry entry;
+ entry.mDelayUs = delayUs;
+ entry.mNumBytes = numBytes;
+ mTotalTransferTimeUs += delayUs;
+ mTotalTransferBytes += numBytes;
+ mBandwidthHistory.push_back(entry);
+
+ // trim old samples, keeping at least kMaxBandwidthHistoryItems samples,
+ // and total transfer time at least kMaxBandwidthHistoryWindowUs.
+ while (mBandwidthHistory.size() > kMaxBandwidthHistoryItems) {
+ List<BandwidthEntry>::iterator it = mBandwidthHistory.begin();
+ if (mTotalTransferTimeUs - it->mDelayUs < kMaxBandwidthHistoryWindowUs) {
+ break;
+ }
+ mTotalTransferTimeUs -= it->mDelayUs;
+ mTotalTransferBytes -= it->mNumBytes;
+ mBandwidthHistory.erase(mBandwidthHistory.begin());
+ }
+}
+
+bool LiveSession::BandwidthEstimator::estimateBandwidth(int32_t *bandwidthBps) {
+ AutoMutex autoLock(mLock);
+
+ if (mBandwidthHistory.size() < 2) {
+ return false;
+ }
+
+ *bandwidthBps = ((double)mTotalTransferBytes * 8E6 / mTotalTransferTimeUs);
+ return true;
+}
+
LiveSession::LiveSession(
const sp<AMessage> ¬ify, uint32_t flags,
const sp<IMediaHTTPService> &httpService)
@@ -66,10 +128,10 @@
mInPreparationPhase(true),
mHTTPDataSource(new MediaHTTP(mHTTPService->makeHTTPConnection())),
mCurBandwidthIndex(-1),
+ mBandwidthEstimator(new BandwidthEstimator()),
mStreamMask(0),
mNewStreamMask(0),
mSwapMask(0),
- mCheckBandwidthGeneration(0),
mSwitchGeneration(0),
mSubtitleGeneration(0),
mLastDequeuedTimeUs(0ll),
@@ -89,13 +151,6 @@
mPacketSources.add(indexToType(i), new AnotherPacketSource(NULL /* meta */));
mPacketSources2.add(indexToType(i), new AnotherPacketSource(NULL /* meta */));
}
-
- size_t numHistoryItems = kBandwidthHistoryBytes /
- PlaylistFetcher::kDownloadBlockSize + 1;
- if (numHistoryItems < 5) {
- numHistoryItems = 5;
- }
- mHTTPDataSource->setBandwidthHistorySize(numHistoryItems);
}
LiveSession::~LiveSession() {
@@ -948,8 +1003,15 @@
}
#endif
-size_t LiveSession::getBandwidthIndex() {
- if (mBandwidthItems.size() == 0) {
+void LiveSession::addBandwidthMeasurement(size_t numBytes, int64_t delayUs) {
+ mBandwidthEstimator->addBandwidthMeasurement(numBytes, delayUs);
+}
+
+size_t LiveSession::getBandwidthIndex(int32_t bandwidthBps) {
+ if (mBandwidthItems.size() < 2) {
+ // shouldn't be here if we only have 1 bandwidth, check
+ // logic to get rid of redundant bandwidth polling
+ ALOGW("getBandwidthIndex() called for single bandwidth playlist!");
return 0;
}
@@ -967,15 +1029,6 @@
}
if (index < 0) {
- int32_t bandwidthBps;
- if (mHTTPDataSource != NULL
- && mHTTPDataSource->estimateBandwidth(&bandwidthBps)) {
- ALOGV("bandwidth estimated at %.2f kbps", bandwidthBps / 1024.0f);
- } else {
- ALOGV("no bandwidth estimate.");
- return 0; // Pick the lowest bandwidth stream by default.
- }
-
char value[PROPERTY_VALUE_MAX];
if (property_get("media.httplive.max-bw", value, NULL)) {
char *end;
@@ -992,15 +1045,9 @@
index = mBandwidthItems.size() - 1;
while (index > 0) {
- // consider only 80% of the available bandwidth, but if we are switching up,
- // be even more conservative (70%) to avoid overestimating and immediately
- // switching back.
- size_t adjustedBandwidthBps = bandwidthBps;
- if (index > mCurBandwidthIndex) {
- adjustedBandwidthBps = adjustedBandwidthBps * 7 / 10;
- } else {
- adjustedBandwidthBps = adjustedBandwidthBps * 8 / 10;
- }
+ // be conservative (70%) to avoid overestimating and immediately
+ // switching down again.
+ size_t adjustedBandwidthBps = bandwidthBps * 7 / 10;
if (mBandwidthItems.itemAt(index).mBandwidth <= adjustedBandwidthBps) {
break;
}
@@ -1577,9 +1624,9 @@
void LiveSession::onPollBuffering() {
ALOGV("onPollBuffering: mSwitchInProgress %d, mReconfigurationInProgress %d, "
- "mInPreparationPhase %d, mStreamMask 0x%x",
+ "mInPreparationPhase %d, mCurBandwidthIndex %d, mStreamMask 0x%x",
mSwitchInProgress, mReconfigurationInProgress,
- mInPreparationPhase, mStreamMask);
+ mInPreparationPhase, mCurBandwidthIndex, mStreamMask);
bool low, mid, high;
if (checkBuffering(low, mid, high)) {
@@ -1588,8 +1635,8 @@
}
// don't switch before we report prepared
- if (!mInPreparationPhase && (low || high)) {
- switchBandwidthIfNeeded(high);
+ if (!mInPreparationPhase) {
+ switchBandwidthIfNeeded(high, !mid);
}
}
@@ -1704,11 +1751,35 @@
return false;
}
-void LiveSession::switchBandwidthIfNeeded(bool canSwitchUp) {
- ssize_t bandwidthIndex = getBandwidthIndex();
+void LiveSession::switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow) {
+ // no need to check bandwidth if we only have 1 bandwidth settings
+ if (mBandwidthItems.size() < 2) {
+ return;
+ }
- if ((canSwitchUp && bandwidthIndex > mCurBandwidthIndex)
- || (!canSwitchUp && bandwidthIndex < mCurBandwidthIndex)) {
+ int32_t bandwidthBps;
+ if (mBandwidthEstimator->estimateBandwidth(&bandwidthBps)) {
+ ALOGV("bandwidth estimated at %.2f kbps", bandwidthBps / 1024.0f);
+ } else {
+ ALOGV("no bandwidth estimate.");
+ return;
+ }
+
+ int32_t curBandwidth = mBandwidthItems.itemAt(mCurBandwidthIndex).mBandwidth;
+ bool bandwidthLow = bandwidthBps < (int32_t)curBandwidth * 8 / 10;
+ bool bandwidthHigh = bandwidthBps > (int32_t)curBandwidth * 12 / 10;
+
+ if ((bufferHigh && bandwidthHigh) || (bufferLow && bandwidthLow)) {
+ ssize_t bandwidthIndex = getBandwidthIndex(bandwidthBps);
+
+ if (bandwidthIndex == mCurBandwidthIndex
+ || (bufferHigh && bandwidthIndex < mCurBandwidthIndex)
+ || (bufferLow && bandwidthIndex > mCurBandwidthIndex)) {
+ return;
+ }
+
+ ALOGI("#### Initiate Bandwidth Switch: %d => %d",
+ mCurBandwidthIndex, bandwidthIndex);
changeConfiguration(-1, bandwidthIndex, false);
}
}
diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h
index 3b0a9a4..685fefa 100644
--- a/media/libstagefright/httplive/LiveSession.h
+++ b/media/libstagefright/httplive/LiveSession.h
@@ -106,7 +106,6 @@
kWhatDisconnect = 'disc',
kWhatSeek = 'seek',
kWhatFetcherNotify = 'notf',
- kWhatCheckBandwidth = 'bndw',
kWhatChangeConfiguration = 'chC0',
kWhatChangeConfiguration2 = 'chC2',
kWhatChangeConfiguration3 = 'chC3',
@@ -115,11 +114,11 @@
kWhatPollBuffering = 'poll',
};
- static const size_t kBandwidthHistoryBytes;
static const int64_t kHighWaterMark;
static const int64_t kMidWaterMark;
static const int64_t kLowWaterMark;
+ struct BandwidthEstimator;
struct BandwidthItem {
size_t mPlaylistIndex;
unsigned long mBandwidth;
@@ -170,6 +169,7 @@
Vector<BandwidthItem> mBandwidthItems;
ssize_t mCurBandwidthIndex;
+ sp<BandwidthEstimator> mBandwidthEstimator;
sp<M3UParser> mPlaylist;
@@ -195,7 +195,6 @@
// * a forced bandwidth switch termination in cancelSwitch on the live looper.
Mutex mSwapMutex;
- int32_t mCheckBandwidthGeneration;
int32_t mSwitchGeneration;
int32_t mSubtitleGeneration;
@@ -249,7 +248,8 @@
sp<M3UParser> fetchPlaylist(
const char *url, uint8_t *curPlaylistHash, bool *unchanged);
- size_t getBandwidthIndex();
+ void addBandwidthMeasurement(size_t numBytes, int64_t delayUs);
+ size_t getBandwidthIndex(int32_t bandwidthBps);
int64_t latestMediaSegmentStartTimeUs();
static int SortByBandwidth(const BandwidthItem *, const BandwidthItem *);
@@ -273,7 +273,7 @@
void cancelPollBuffering();
void onPollBuffering();
bool checkBuffering(bool &low, bool &mid, bool &high);
- void switchBandwidthIfNeeded(bool canSwitchUp);
+ void switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow);
void finishDisconnect();
diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index 7f818a8..a447010 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -648,23 +648,23 @@
targetDurationUs = targetDurationSecs * 1000000ll;
}
- int64_t durationToBufferUs = kMinBufferedDurationUs;
-
int64_t bufferedDurationUs = 0ll;
- status_t finalResult = NOT_ENOUGH_DATA;
+ status_t finalResult = OK;
if (mStreamTypeMask == LiveSession::STREAMTYPE_SUBTITLES) {
sp<AnotherPacketSource> packetSource =
mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES);
bufferedDurationUs =
packetSource->getBufferedDurationUs(&finalResult);
- finalResult = OK;
} else {
- // Use max stream duration to prevent us from waiting on a non-existent stream;
- // when we cannot make out from the manifest what streams are included in a playlist
- // we might assume extra streams.
+ // Use min stream duration, but ignore streams that never have any packet
+ // enqueued to prevent us from waiting on a non-existent stream;
+ // when we cannot make out from the manifest what streams are included in
+ // a playlist we might assume extra streams.
+ bufferedDurationUs = -1ll;
for (size_t i = 0; i < mPacketSources.size(); ++i) {
- if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0) {
+ if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0
+ || mPacketSources[i]->getLatestEnqueuedMeta() == NULL) {
continue;
}
@@ -672,24 +672,19 @@
mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult);
ALOGV("buffered %" PRId64 " for stream %d",
bufferedStreamDurationUs, mPacketSources.keyAt(i));
- if (bufferedStreamDurationUs > bufferedDurationUs) {
+ if (bufferedDurationUs == -1ll
+ || bufferedStreamDurationUs < bufferedDurationUs) {
bufferedDurationUs = bufferedStreamDurationUs;
}
}
- }
- downloadMore = (bufferedDurationUs < durationToBufferUs);
-
- // signal start if buffered up at least the target size
- if (!mPrepared && bufferedDurationUs > targetDurationUs && downloadMore) {
- mPrepared = true;
-
- ALOGV("prepared, buffered=%" PRId64 " > %" PRId64 "",
- bufferedDurationUs, targetDurationUs);
+ if (bufferedDurationUs == -1ll) {
+ bufferedDurationUs = 0ll;
+ }
}
- if (finalResult == OK && downloadMore) {
+ if (finalResult == OK && bufferedDurationUs < kMinBufferedDurationUs) {
ALOGV("monitoring, buffered=%" PRId64 " < %" PRId64 "",
- bufferedDurationUs, durationToBufferUs);
+ bufferedDurationUs, kMinBufferedDurationUs);
// delay the next download slightly; hopefully this gives other concurrent fetchers
// a better chance to run.
// onDownloadNext();
@@ -697,13 +692,16 @@
msg->setInt32("generation", mMonitorQueueGeneration);
msg->post(1000l);
} else {
- // Nothing to do yet, try again in a second.
- int64_t delayUs = mPrepared ? kMaxMonitorDelayUs : targetDurationUs / 2;
+ // We'd like to maintain buffering above durationToBufferUs, so try
+ // again when buffer just about to go below durationToBufferUs
+ // (or after targetDurationUs / 2, whichever is smaller).
+ int64_t delayUs = bufferedDurationUs - kMinBufferedDurationUs + 1000000ll;
+ if (delayUs > targetDurationUs / 2) {
+ delayUs = targetDurationUs / 2;
+ }
ALOGV("pausing for %" PRId64 ", buffered=%" PRId64 " > %" PRId64 "",
- delayUs, bufferedDurationUs, durationToBufferUs);
- // :TRICKY: need to enforce minimum delay because the delay to
- // refresh the playlist will become 0
- postMonitorQueue(delayUs, mPrepared ? targetDurationUs * 2 : 0);
+ delayUs, bufferedDurationUs, kMinBufferedDurationUs);
+ postMonitorQueue(delayUs);
}
}
@@ -986,9 +984,20 @@
// block-wise download
ssize_t bytesRead;
do {
+ int64_t startUs = ALooper::GetNowUs();
+
bytesRead = mSession->fetchFile(
uri.c_str(), &buffer, range_offset, range_length, kDownloadBlockSize, &source);
+ // add sample for bandwidth estimation (excluding subtitles)
+ if (bytesRead > 0
+ && (mStreamTypeMask
+ & (LiveSession::STREAMTYPE_AUDIO
+ | LiveSession::STREAMTYPE_VIDEO))) {
+ int64_t delayUs = ALooper::GetNowUs() - startUs;
+ mSession->addBandwidthMeasurement(bytesRead, delayUs);
+ }
+
if (bytesRead < 0) {
status_t err = bytesRead;
ALOGE("failed to fetch .ts segment at url '%s'", uri.c_str());