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-rw-r--r--media/libstagefright/httplive/LiveSession.cpp214
1 files changed, 140 insertions, 74 deletions
diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index 2d93152..74f58e9 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -141,6 +141,21 @@ const char *LiveSession::getKeyForStream(StreamType type) {
return NULL;
}
+//static
+const char *LiveSession::getNameForStream(StreamType type) {
+ switch (type) {
+ case STREAMTYPE_VIDEO:
+ return "video";
+ case STREAMTYPE_AUDIO:
+ return "audio";
+ case STREAMTYPE_SUBTITLES:
+ return "subs";
+ default:
+ break;
+ }
+ return "unknown";
+}
+
LiveSession::LiveSession(
const sp<AMessage> &notify, uint32_t flags,
const sp<IMediaHTTPService> &httpService)
@@ -192,7 +207,11 @@ status_t LiveSession::dequeueAccessUnit(
status_t finalResult = OK;
sp<AnotherPacketSource> packetSource = mPacketSources.valueFor(stream);
- ssize_t idx = typeToIndex(stream);
+ ssize_t streamIdx = typeToIndex(stream);
+ if (streamIdx < 0) {
+ return INVALID_VALUE;
+ }
+ const char *streamStr = getNameForStream(stream);
// Do not let client pull data if we don't have data packets yet.
// We might only have a format discontinuity queued without data.
// When NuPlayerDecoder dequeues the format discontinuity, it will
@@ -200,6 +219,9 @@ status_t LiveSession::dequeueAccessUnit(
// thinks it can do seamless change, so will not shutdown decoder.
// When the actual format arrives, it can't handle it and get stuck.
if (!packetSource->hasDataBufferAvailable(&finalResult)) {
+ ALOGV("[%s] dequeueAccessUnit: no buffer available (finalResult=%d)",
+ streamStr, finalResult);
+
if (finalResult == OK) {
return -EAGAIN;
} else {
@@ -212,25 +234,6 @@ status_t LiveSession::dequeueAccessUnit(
status_t err = packetSource->dequeueAccessUnit(accessUnit);
- size_t streamIdx;
- const char *streamStr;
- switch (stream) {
- case STREAMTYPE_AUDIO:
- streamIdx = kAudioIndex;
- streamStr = "audio";
- break;
- case STREAMTYPE_VIDEO:
- streamIdx = kVideoIndex;
- streamStr = "video";
- break;
- case STREAMTYPE_SUBTITLES:
- streamIdx = kSubtitleIndex;
- streamStr = "subs";
- break;
- default:
- TRESPASS();
- }
-
StreamItem& strm = mStreams[streamIdx];
if (err == INFO_DISCONTINUITY) {
// adaptive streaming, discontinuities in the playlist
@@ -249,9 +252,10 @@ status_t LiveSession::dequeueAccessUnit(
} else if (err == OK) {
if (stream == STREAMTYPE_AUDIO || stream == STREAMTYPE_VIDEO) {
- int64_t timeUs;
+ int64_t timeUs, originalTimeUs;
int32_t discontinuitySeq = 0;
CHECK((*accessUnit)->meta()->findInt64("timeUs", &timeUs));
+ originalTimeUs = timeUs;
(*accessUnit)->meta()->findInt32("discontinuitySeq", &discontinuitySeq);
if (discontinuitySeq > (int32_t) strm.mCurDiscontinuitySeq) {
int64_t offsetTimeUs;
@@ -303,7 +307,8 @@ status_t LiveSession::dequeueAccessUnit(
timeUs += mDiscontinuityOffsetTimesUs.valueFor(discontinuitySeq);
}
- ALOGV("[%s] read buffer at time %" PRId64 " us", streamStr, timeUs);
+ ALOGV("[%s] dequeueAccessUnit: time %lld us, original %lld us",
+ streamStr, (long long)timeUs, (long long)originalTimeUs);
(*accessUnit)->meta()->setInt64("timeUs", timeUs);
mLastDequeuedTimeUs = timeUs;
mRealTimeBaseUs = ALooper::GetNowUs() - timeUs;
@@ -409,7 +414,7 @@ bool LiveSession::checkSwitchProgress(
if (lastDequeueMeta == NULL) {
// this means we don't have enough cushion, try again later
ALOGV("[%s] up switching failed due to insufficient buffer",
- stream == STREAMTYPE_AUDIO ? "audio" : "video");
+ getNameForStream(stream));
return false;
}
} else {
@@ -428,7 +433,7 @@ bool LiveSession::checkSwitchProgress(
if (firstNewMeta[i] == NULL) {
HLSTime dequeueTime(lastDequeueMeta);
ALOGV("[%s] dequeue time (%d, %lld) past start time",
- stream == STREAMTYPE_AUDIO ? "audio" : "video",
+ getNameForStream(stream),
dequeueTime.mSeq, (long long) dequeueTime.mTimeUs);
return false;
}
@@ -493,16 +498,15 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) {
case kWhatSeek:
{
- sp<AReplyToken> seekReplyID;
- CHECK(msg->senderAwaitsResponse(&seekReplyID));
- mSeekReplyID = seekReplyID;
- mSeekReply = new AMessage;
-
- status_t err = onSeek(msg);
-
- if (err != OK) {
+ if (mReconfigurationInProgress) {
msg->post(50000);
+ break;
}
+
+ CHECK(msg->senderAwaitsResponse(&mSeekReplyID));
+ mSeekReply = new AMessage;
+
+ onSeek(msg);
break;
}
@@ -525,6 +529,11 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) {
break;
}
+ ALOGV("fetcher-%d %s",
+ mFetcherInfos[index].mFetcher->getFetcherID(),
+ what == PlaylistFetcher::kWhatPaused ?
+ "paused" : "stopped");
+
if (what == PlaylistFetcher::kWhatStopped) {
mFetcherLooper->unregisterHandler(
mFetcherInfos[index].mFetcher->id());
@@ -544,6 +553,7 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) {
if (--mContinuationCounter == 0) {
mContinuation->post();
}
+ ALOGV("%zu fetcher(s) left", mContinuationCounter);
}
break;
}
@@ -636,6 +646,9 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) {
int32_t switchGeneration;
CHECK(msg->findInt32("switchGeneration", &switchGeneration));
+ ALOGV("kWhatStartedAt: switchGen=%d, mSwitchGen=%d",
+ switchGeneration, mSwitchGeneration);
+
if (switchGeneration != mSwitchGeneration) {
break;
}
@@ -667,6 +680,7 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) {
if (checkSwitchProgress(stopParams, delayUs, &needResumeUntil)) {
// playback time hasn't passed startAt time
if (!needResumeUntil) {
+ ALOGV("finish switch");
for (size_t i = 0; i < kMaxStreams; ++i) {
if ((mSwapMask & indexToType(i))
&& uri == mStreams[i].mNewUri) {
@@ -682,6 +696,7 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) {
// Resume fetcher for the original variant; the resumed fetcher should
// continue until the timestamps found in msg, which is stored by the
// new fetcher to indicate where the new variant has started buffering.
+ ALOGV("finish switch with resumeUntilAsync");
for (size_t i = 0; i < mFetcherInfos.size(); i++) {
const FetcherInfo &info = mFetcherInfos.valueAt(i);
if (info.mToBeRemoved) {
@@ -693,8 +708,10 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) {
// playback time passed startAt time
if (switchUp) {
// if switching up, cancel and retry if condition satisfies again
+ ALOGV("cancel up switch because we're too late");
cancelBandwidthSwitch(true /* resume */);
} else {
+ ALOGV("retry down switch at next sample");
resumeFetcher(uri, mSwapMask, -1, true /* newUri */);
}
}
@@ -933,7 +950,8 @@ sp<PlaylistFetcher> LiveSession::addFetcher(const char *uri) {
notify->setInt32("switchGeneration", mSwitchGeneration);
FetcherInfo info;
- info.mFetcher = new PlaylistFetcher(notify, this, uri, mSubtitleGeneration);
+ info.mFetcher = new PlaylistFetcher(
+ notify, this, uri, mCurBandwidthIndex, mSubtitleGeneration);
info.mDurationUs = -1ll;
info.mToBeRemoved = false;
info.mToBeResumed = false;
@@ -1167,9 +1185,13 @@ bool LiveSession::resumeFetcher(
}
if (resume) {
- ALOGV("resuming fetcher %s, timeUs %lld", uri.c_str(), (long long)timeUs);
+ sp<PlaylistFetcher> &fetcher = mFetcherInfos.editValueAt(index).mFetcher;
SeekMode seekMode = newUri ? kSeekModeNextSample : kSeekModeExactPosition;
- mFetcherInfos.editValueAt(index).mFetcher->startAsync(
+
+ ALOGV("resuming fetcher-%d, timeUs=%lld, seekMode=%d",
+ fetcher->getFetcherID(), (long long)timeUs, seekMode);
+
+ fetcher->startAsync(
sources[kAudioIndex],
sources[kVideoIndex],
sources[kSubtitleIndex],
@@ -1349,16 +1371,10 @@ HLSTime LiveSession::latestMediaSegmentStartTime() const {
return audioTime < videoTime ? videoTime : audioTime;
}
-status_t LiveSession::onSeek(const sp<AMessage> &msg) {
+void LiveSession::onSeek(const sp<AMessage> &msg) {
int64_t timeUs;
CHECK(msg->findInt64("timeUs", &timeUs));
-
- if (!mReconfigurationInProgress) {
- changeConfiguration(timeUs);
- return OK;
- } else {
- return -EWOULDBLOCK;
- }
+ changeConfiguration(timeUs);
}
status_t LiveSession::getDuration(int64_t *durationUs) const {
@@ -1406,6 +1422,9 @@ status_t LiveSession::selectTrack(size_t index, bool select) {
return INVALID_OPERATION;
}
+ ALOGV("selectTrack: index=%zu, select=%d, mSubtitleGen=%d++",
+ index, select, mSubtitleGeneration);
+
++mSubtitleGeneration;
status_t err = mPlaylist->selectTrack(index, select);
if (err == OK) {
@@ -1426,6 +1445,9 @@ ssize_t LiveSession::getSelectedTrack(media_track_type type) const {
void LiveSession::changeConfiguration(
int64_t timeUs, ssize_t bandwidthIndex, bool pickTrack) {
+ ALOGV("changeConfiguration: timeUs=%lld us, bwIndex=%zd, pickTrack=%d",
+ (long long)timeUs, bandwidthIndex, pickTrack);
+
cancelBandwidthSwitch();
CHECK(!mReconfigurationInProgress);
@@ -1433,6 +1455,10 @@ void LiveSession::changeConfiguration(
if (bandwidthIndex >= 0) {
mOrigBandwidthIndex = mCurBandwidthIndex;
mCurBandwidthIndex = bandwidthIndex;
+ if (mOrigBandwidthIndex != mCurBandwidthIndex) {
+ ALOGI("#### Starting Bandwidth Switch: %zd => %zd",
+ mOrigBandwidthIndex, mCurBandwidthIndex);
+ }
}
CHECK_LT(mCurBandwidthIndex, mBandwidthItems.size());
const BandwidthItem &item = mBandwidthItems.itemAt(mCurBandwidthIndex);
@@ -1478,6 +1504,7 @@ void LiveSession::changeConfiguration(
}
if (discardFetcher) {
+ ALOGV("discarding fetcher-%d", fetcher->getFetcherID());
fetcher->stopAsync();
} else {
float threshold = -1.0f; // always finish fetching by default
@@ -1490,8 +1517,8 @@ void LiveSession::changeConfiguration(
mOrigBandwidthIndex, mCurBandwidthIndex);
}
- ALOGV("Pausing with threshold %.3f", threshold);
-
+ ALOGV("pausing fetcher-%d, threshold=%.2f",
+ fetcher->getFetcherID(), threshold);
fetcher->pauseAsync(threshold);
}
}
@@ -1526,6 +1553,8 @@ void LiveSession::changeConfiguration(
}
void LiveSession::onChangeConfiguration(const sp<AMessage> &msg) {
+ ALOGV("onChangeConfiguration");
+
if (!mReconfigurationInProgress) {
int32_t pickTrack = 0;
msg->findInt32("pickTrack", &pickTrack);
@@ -1536,6 +1565,8 @@ void LiveSession::onChangeConfiguration(const sp<AMessage> &msg) {
}
void LiveSession::onChangeConfiguration2(const sp<AMessage> &msg) {
+ ALOGV("onChangeConfiguration2");
+
mContinuation.clear();
// All fetchers are either suspended or have been removed now.
@@ -1547,6 +1578,7 @@ void LiveSession::onChangeConfiguration2(const sp<AMessage> &msg) {
if (timeUs >= 0) {
mLastSeekTimeUs = timeUs;
+ mLastDequeuedTimeUs = timeUs;
for (size_t i = 0; i < mPacketSources.size(); i++) {
mPacketSources.editValueAt(i)->clear();
@@ -1599,8 +1631,10 @@ void LiveSession::onChangeConfiguration2(const sp<AMessage> &msg) {
ALOGV("stream %zu changed: oldURI %s, newURI %s", i,
mStreams[i].mUri.c_str(), URIs[i].c_str());
sp<AnotherPacketSource> source = mPacketSources.valueFor(indexToType(i));
- source->queueDiscontinuity(
- ATSParser::DISCONTINUITY_FORMATCHANGE, NULL, true);
+ if (source->getLatestDequeuedMeta() != NULL) {
+ source->queueDiscontinuity(
+ ATSParser::DISCONTINUITY_FORMATCHANGE, NULL, true);
+ }
}
// Determine which decoders to shutdown on the player side,
// a decoder has to be shutdown if its streamtype was active
@@ -1660,16 +1694,17 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
// and resume audio.
mSwapMask = mNewStreamMask & mStreamMask & ~resumeMask;
switching = (mSwapMask != 0);
- if (!switching) {
- ALOGV("#### Finishing Bandwidth Switch Early: %zd => %zd",
- mOrigBandwidthIndex, mCurBandwidthIndex);
- }
}
mRealTimeBaseUs = ALooper::GetNowUs() - mLastDequeuedTimeUs;
} else {
mRealTimeBaseUs = ALooper::GetNowUs() - timeUs;
}
+ ALOGV("onChangeConfiguration3: timeUs=%lld, switching=%d, pickTrack=%d, "
+ "mStreamMask=0x%x, mNewStreamMask=0x%x, mSwapMask=0x%x",
+ (long long)timeUs, switching, pickTrack,
+ mStreamMask, mNewStreamMask, mSwapMask);
+
for (size_t i = 0; i < kMaxStreams; ++i) {
if (streamMask & indexToType(i)) {
if (switching) {
@@ -1687,6 +1722,9 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
const AString &uri = mFetcherInfos.keyAt(i);
if (!resumeFetcher(uri, resumeMask, timeUs)) {
+ ALOGV("marking fetcher-%d to be removed",
+ mFetcherInfos[i].mFetcher->getFetcherID());
+
mFetcherInfos.editValueAt(i).mToBeRemoved = true;
}
}
@@ -1776,6 +1814,14 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
}
}
+ ALOGV("[fetcher-%d] startAsync: startTimeUs %lld mLastSeekTimeUs %lld "
+ "segmentStartTimeUs %lld seekMode %d",
+ fetcher->getFetcherID(),
+ (long long)startTime.mTimeUs,
+ (long long)mLastSeekTimeUs,
+ (long long)startTime.getSegmentTimeUs(true /* midpoint */),
+ seekMode);
+
// Set the target segment start time to the middle point of the
// segment where the last sample was.
// This gives a better guess if segments of the two variants are not
@@ -1795,22 +1841,28 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
// All fetchers have now been started, the configuration change
// has completed.
- ALOGV("XXX configuration change completed.");
mReconfigurationInProgress = false;
if (switching) {
mSwitchInProgress = true;
} else {
mStreamMask = mNewStreamMask;
- mOrigBandwidthIndex = mCurBandwidthIndex;
+ if (mOrigBandwidthIndex != mCurBandwidthIndex) {
+ ALOGV("#### Finished Bandwidth Switch Early: %zd => %zd",
+ mOrigBandwidthIndex, mCurBandwidthIndex);
+ mOrigBandwidthIndex = mCurBandwidthIndex;
+ }
}
+ ALOGV("onChangeConfiguration3: mSwitchInProgress %d, mStreamMask 0x%x",
+ mSwitchInProgress, mStreamMask);
+
if (mDisconnectReplyID != NULL) {
finishDisconnect();
}
}
void LiveSession::swapPacketSource(StreamType stream) {
- ALOGV("swapPacketSource: stream = %d", stream);
+ ALOGV("[%s] swapPacketSource", getNameForStream(stream));
// transfer packets from source2 to source
sp<AnotherPacketSource> &aps = mPacketSources.editValueFor(stream);
@@ -1858,7 +1910,7 @@ void LiveSession::tryToFinishBandwidthSwitch(const AString &oldUri) {
mFetcherInfos.editValueAt(index).mFetcher->stopAsync(false /* clear */);
- ALOGV("tryToFinishBandwidthSwitch: mSwapMask=%x", mSwapMask);
+ ALOGV("tryToFinishBandwidthSwitch: mSwapMask=0x%x", mSwapMask);
if (mSwapMask != 0) {
return;
}
@@ -1925,11 +1977,19 @@ void LiveSession::onPollBuffering() {
bool underflow, ready, down, up;
if (checkBuffering(underflow, ready, down, up)) {
- if (mInPreparationPhase && ready) {
- postPrepared(OK);
+ if (mInPreparationPhase) {
+ // Allow down switch even if we're still preparing.
+ //
+ // Some streams have a high bandwidth index as default,
+ // when bandwidth is low, it takes a long time to buffer
+ // to ready mark, then it immediately pauses after start
+ // as we have to do a down switch. It's better experience
+ // to restart from a lower index, if we detect low bw.
+ if (!switchBandwidthIfNeeded(false /* up */, down) && ready) {
+ postPrepared(OK);
+ }
}
- // don't switch before we report prepared
if (!mInPreparationPhase) {
if (ready) {
stopBufferingIfNecessary();
@@ -1937,8 +1997,7 @@ void LiveSession::onPollBuffering() {
startBufferingIfNecessary();
}
switchBandwidthIfNeeded(up, down);
- }
-
+ }
}
schedulePollBuffering();
@@ -1983,7 +2042,7 @@ void LiveSession::cancelBandwidthSwitch(bool resume) {
}
ALOGI("#### Canceled Bandwidth Switch: %zd => %zd",
- mCurBandwidthIndex, mOrigBandwidthIndex);
+ mOrigBandwidthIndex, mCurBandwidthIndex);
mSwitchGeneration++;
mSwitchInProgress = false;
@@ -2022,13 +2081,16 @@ bool LiveSession::checkBuffering(
int64_t bufferedDurationUs =
mPacketSources[i]->getEstimatedDurationUs();
- ALOGV("source[%zu]: buffered %lld us", i, (long long)bufferedDurationUs);
+ ALOGV("[%s] buffered %lld us",
+ getNameForStream(mPacketSources.keyAt(i)),
+ (long long)bufferedDurationUs);
if (durationUs >= 0) {
int32_t percent;
if (mPacketSources[i]->isFinished(0 /* duration */)) {
percent = 100;
} else {
- percent = (int32_t)(100.0 * (mLastDequeuedTimeUs + bufferedDurationUs) / durationUs);
+ percent = (int32_t)(100.0 *
+ (mLastDequeuedTimeUs + bufferedDurationUs) / durationUs);
}
if (minBufferPercent < 0 || percent < minBufferPercent) {
minBufferPercent = percent;
@@ -2111,10 +2173,14 @@ void LiveSession::notifyBufferingUpdate(int32_t percentage) {
notify->post();
}
-void LiveSession::switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow) {
+/*
+ * returns true if a bandwidth switch is actually needed (and started),
+ * returns false otherwise
+ */
+bool LiveSession::switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow) {
// no need to check bandwidth if we only have 1 bandwidth settings
if (mSwitchInProgress || mBandwidthItems.size() < 2) {
- return;
+ return false;
}
int32_t bandwidthBps;
@@ -2123,7 +2189,7 @@ void LiveSession::switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow) {
mLastBandwidthBps = bandwidthBps;
} else {
ALOGV("no bandwidth estimate.");
- return;
+ return false;
}
int32_t curBandwidth = mBandwidthItems.itemAt(mCurBandwidthIndex).mBandwidth;
@@ -2142,16 +2208,16 @@ void LiveSession::switchBandwidthIfNeeded(bool bufferHigh, bool bufferLow) {
// bandwidthIndex is < mCurBandwidthIndex, as getBandwidthIndex() only uses 70%
// of measured bw. In that case we don't want to do anything, since we have
// both enough buffer and enough bw.
- if (bandwidthIndex == mCurBandwidthIndex
- || (canSwitchUp && bandwidthIndex < mCurBandwidthIndex)
- || (canSwithDown && bandwidthIndex > mCurBandwidthIndex)) {
- return;
+ if ((canSwitchUp && bandwidthIndex > mCurBandwidthIndex)
+ || (canSwithDown && bandwidthIndex < mCurBandwidthIndex)) {
+ // if not yet prepared, just restart again with new bw index.
+ // this is faster and playback experience is cleaner.
+ changeConfiguration(
+ mInPreparationPhase ? 0 : -1ll, bandwidthIndex);
+ return true;
}
-
- ALOGI("#### Starting Bandwidth Switch: %zd => %zd",
- mCurBandwidthIndex, bandwidthIndex);
- changeConfiguration(-1, bandwidthIndex, false);
}
+ return false;
}
void LiveSession::postError(status_t err) {
generated by cgit v1.1 at 2024-05-31 16:28:43 +0000