From 4dd0a8a3d66c2853faf2834565b3c5df4f68734d Mon Sep 17 00:00:00 2001
From: Lajos Molnar <lajos@google.com>
Date: Wed, 6 Nov 2013 21:04:34 -0800
Subject: WA: Queue extra buffers on output port during reconfig if input
 EOS-ed

Some codecs may return input buffers before having them processed.
This causes a halt if we already signaled an EOS on the input
port.  For now keep submitting output meta buffers one at a time if
the input EOS-ed, but the output not yet.

Normally, we submit an output buffer for each input buffer that is
with the component (waiting to be processed).

Change-Id: I8a1251bfb504f40f1e4085a1e220bf9a4d0b05d9
Signed-off-by: Lajos Molnar <lajos@google.com>
Bug: 11433909
---
 media/libstagefright/ACodec.cpp | 28 ++++++++++++++++++++++++++++
 1 file changed, 28 insertions(+)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 528fdb9..5d5220f 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -452,6 +452,18 @@ void ACodec::signalRequestIDRFrame() {
     (new AMessage(kWhatRequestIDRFrame, id()))->post();
 }
 
+// *** NOTE: THE FOLLOWING WORKAROUND WILL BE REMOVED ***
+// Some codecs may return input buffers before having them processed.
+// This causes a halt if we already signaled an EOS on the input
+// port.  For now keep submitting an output buffer if there was an
+// EOS on the input port, but not yet on the output port.
+void ACodec::signalSubmitOutputMetaDataBufferIfEOS_workaround() {
+    if (mPortEOS[kPortIndexInput] && !mPortEOS[kPortIndexOutput] &&
+            mMetaDataBuffersToSubmit > 0) {
+        (new AMessage(kWhatSubmitOutputMetaDataBufferIfEOS, id()))->post();
+    }
+}
+
 status_t ACodec::allocateBuffersOnPort(OMX_U32 portIndex) {
     CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput);
 
@@ -4036,6 +4048,9 @@ void ACodec::ExecutingState::submitOutputMetaBuffers() {
                 break;
         }
     }
+
+    // *** NOTE: THE FOLLOWING WORKAROUND WILL BE REMOVED ***
+    mCodec->signalSubmitOutputMetaDataBufferIfEOS_workaround();
 }
 
 void ACodec::ExecutingState::submitRegularOutputBuffers() {
@@ -4184,6 +4199,19 @@ bool ACodec::ExecutingState::onMessageReceived(const sp<AMessage> &msg) {
             break;
         }
 
+        // *** NOTE: THE FOLLOWING WORKAROUND WILL BE REMOVED ***
+        case kWhatSubmitOutputMetaDataBufferIfEOS:
+        {
+            if (mCodec->mPortEOS[kPortIndexInput] &&
+                    !mCodec->mPortEOS[kPortIndexOutput]) {
+                status_t err = mCodec->submitOutputMetaDataBuffer();
+                if (err == OK) {
+                    mCodec->signalSubmitOutputMetaDataBufferIfEOS_workaround();
+                }
+            }
+            return true;
+        }
+
         default:
             handled = BaseState::onMessageReceived(msg);
             break;
-- 
cgit v1.1


From d0115d8c4db2f337030dea706bc0a434c4a03ed6 Mon Sep 17 00:00:00 2001
From: Eric Laurent <elaurent@google.com>
Date: Fri, 26 Jul 2013 17:16:50 -0700
Subject: update offloaded audio track sampling rate

AudioPlayer must read the sampling rate from offloaded audio sinks
whenever a new time position is computed as the decoder can update
the sampling rate on the fly.

Bug: 12823955.

Change-Id: I997e5248cfd4017aeceb4e11689324ded2a5bc88
---
 media/libstagefright/AudioPlayer.cpp | 15 +++++++++++++--
 1 file changed, 13 insertions(+), 2 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index a8a8786..05ee34e 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -721,16 +721,27 @@ int64_t AudioPlayer::getRealTimeUsLocked() const {
     return result + diffUs;
 }
 
-int64_t AudioPlayer::getOutputPlayPositionUs_l() const
+int64_t AudioPlayer::getOutputPlayPositionUs_l()
 {
     uint32_t playedSamples = 0;
+    uint32_t sampleRate;
     if (mAudioSink != NULL) {
         mAudioSink->getPosition(&playedSamples);
+        sampleRate = mAudioSink->getSampleRate();
     } else {
         mAudioTrack->getPosition(&playedSamples);
+        sampleRate = mAudioTrack->getSampleRate();
+    }
+    if (sampleRate != 0) {
+        mSampleRate = sampleRate;
     }
 
-    const int64_t playedUs = (static_cast<int64_t>(playedSamples) * 1000000 ) / mSampleRate;
+    int64_t playedUs;
+    if (mSampleRate != 0) {
+        playedUs = (static_cast<int64_t>(playedSamples) * 1000000 ) / mSampleRate;
+    } else {
+        playedUs = 0;
+    }
 
     // HAL position is relative to the first buffer we sent at mStartPosUs
     const int64_t renderedDuration = mStartPosUs + playedUs;
-- 
cgit v1.1


From 145b727b4cce8b1b8d161612f30d12bbdfc8d134 Mon Sep 17 00:00:00 2001
From: Haynes Mathew George <hgeorge@codeaurora.org>
Date: Fri, 6 Dec 2013 11:31:57 -0800
Subject: libstagefright: Delay release of wakelock in TimedEventQueue

Delay release of wakelock in the TimedEventQueue to
after an event has been processed.
This ensures AP shutdown does not happen while an event
is ready but hasn't been processed yet.

Bug: 11976087.
Change-Id: I9bce83b9ff0f2b4d174a53c95e5b0f135b5ccc70
---
 media/libstagefright/TimedEventQueue.cpp       | 13 ++++++++-----
 media/libstagefright/include/TimedEventQueue.h |  2 +-
 2 files changed, 9 insertions(+), 6 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/TimedEventQueue.cpp b/media/libstagefright/TimedEventQueue.cpp
index 1a9a26b..dedd186 100644
--- a/media/libstagefright/TimedEventQueue.cpp
+++ b/media/libstagefright/TimedEventQueue.cpp
@@ -217,6 +217,7 @@ void TimedEventQueue::threadEntry() {
     for (;;) {
         int64_t now_us = 0;
         sp<Event> event;
+        bool wakeLocked = false;
 
         {
             Mutex::Autolock autoLock(mLock);
@@ -283,26 +284,28 @@ void TimedEventQueue::threadEntry() {
             // removeEventFromQueue_l will return NULL.
             // Otherwise, the QueueItem will be removed
             // from the queue and the referenced event returned.
-            event = removeEventFromQueue_l(eventID);
+            event = removeEventFromQueue_l(eventID, &wakeLocked);
         }
 
         if (event != NULL) {
             // Fire event with the lock NOT held.
             event->fire(this, now_us);
+            if (wakeLocked) {
+                Mutex::Autolock autoLock(mLock);
+                releaseWakeLock_l();
+            }
         }
     }
 }
 
 sp<TimedEventQueue::Event> TimedEventQueue::removeEventFromQueue_l(
-        event_id id) {
+        event_id id, bool *wakeLocked) {
     for (List<QueueItem>::iterator it = mQueue.begin();
          it != mQueue.end(); ++it) {
         if ((*it).event->eventID() == id) {
             sp<Event> event = (*it).event;
             event->setEventID(0);
-            if ((*it).has_wakelock) {
-                releaseWakeLock_l();
-            }
+            *wakeLocked = (*it).has_wakelock;
             mQueue.erase(it);
             return event;
         }
diff --git a/media/libstagefright/include/TimedEventQueue.h b/media/libstagefright/include/TimedEventQueue.h
index 38a08b1..3e84256 100644
--- a/media/libstagefright/include/TimedEventQueue.h
+++ b/media/libstagefright/include/TimedEventQueue.h
@@ -145,7 +145,7 @@ private:
     static void *ThreadWrapper(void *me);
     void threadEntry();
 
-    sp<Event> removeEventFromQueue_l(event_id id);
+    sp<Event> removeEventFromQueue_l(event_id id, bool *wakeLocked);
 
     void acquireWakeLock_l();
     void releaseWakeLock_l(bool force = false);
-- 
cgit v1.1


From 3c836e7882e88de42e0e600627de433b61e78621 Mon Sep 17 00:00:00 2001
From: Eric Laurent <elaurent@google.com>
Date: Fri, 6 Dec 2013 11:16:54 -0800
Subject: stagefright: fix offloading HE-AAC sampling rate.

Fix HE AAC SBR sampling rate reading for explicit
signaling.

Bug: 11697128.
Change-Id: Ifec0ab9d48d9293f6774ec1efd9da9445994cb7c
---
 media/libstagefright/MPEG4Extractor.cpp | 40 +++++++++++++++++++--------------
 1 file changed, 23 insertions(+), 17 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 1ba1c6e..6f51c39 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -2285,6 +2285,11 @@ status_t MPEG4Extractor::updateAudioTrackInfoFromESDS_MPEG4Audio(
         return ERROR_MALFORMED;
     }
 
+    static uint32_t kSamplingRate[] = {
+        96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
+        16000, 12000, 11025, 8000, 7350
+    };
+
     ABitReader br(csd, csd_size);
     uint32_t objectType = br.getBits(5);
 
@@ -2304,29 +2309,30 @@ status_t MPEG4Extractor::updateAudioTrackInfoFromESDS_MPEG4Audio(
         numChannels = br.getBits(4);
     } else {
         numChannels = br.getBits(4);
-        if (objectType == 5) {
-            // SBR specific config per 14496-3 table 1.13
-            freqIndex = br.getBits(4);
-            if (freqIndex == 15) {
-                if (csd_size < 8) {
-                    return ERROR_MALFORMED;
-                }
-                sampleRate = br.getBits(24);
-            }
+
+        if (freqIndex == 13 || freqIndex == 14) {
+            return ERROR_MALFORMED;
         }
 
-        if (sampleRate == 0) {
-            static uint32_t kSamplingRate[] = {
-                96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
-                16000, 12000, 11025, 8000, 7350
-            };
+        sampleRate = kSamplingRate[freqIndex];
+    }
 
-            if (freqIndex == 13 || freqIndex == 14) {
+    if (objectType == 5 || objectType == 29) { // SBR specific config per 14496-3 table 1.13
+        uint32_t extFreqIndex = br.getBits(4);
+        int32_t extSampleRate;
+        if (extFreqIndex == 15) {
+            if (csd_size < 8) {
                 return ERROR_MALFORMED;
             }
-
-            sampleRate = kSamplingRate[freqIndex];
+            extSampleRate = br.getBits(24);
+        } else {
+            if (extFreqIndex == 13 || extFreqIndex == 14) {
+                return ERROR_MALFORMED;
+            }
+            extSampleRate = kSamplingRate[extFreqIndex];
         }
+        //TODO: save the extension sampling rate value in meta data =>
+        //      mLastTrack->meta->setInt32(kKeyExtSampleRate, extSampleRate);
     }
 
     if (numChannels == 0) {
-- 
cgit v1.1


From 3b7ddf83db2d11c08116a6476c0f11e71bd72ba4 Mon Sep 17 00:00:00 2001
From: Chong Zhang <chz@google.com>
Date: Fri, 7 Feb 2014 12:05:34 -0800
Subject: set AAC MPEG version to MPEG-2

wifi display play spec only allows MPEG-2 AAC.

Bug: 12747927
Change-Id: Ic96d27ce57f5c8e7da55e43b38f629b5c7333817
---
 media/libstagefright/wifi-display/source/TSPacketizer.cpp | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/wifi-display/source/TSPacketizer.cpp b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
index c674700..eeb3700 100644
--- a/media/libstagefright/wifi-display/source/TSPacketizer.cpp
+++ b/media/libstagefright/wifi-display/source/TSPacketizer.cpp
@@ -216,7 +216,7 @@ sp<ABuffer> TSPacketizer::Track::prependADTSHeader(
     uint8_t *ptr = dup->data();
 
     *ptr++ = 0xff;
-    *ptr++ = 0xf1;  // b11110001, ID=0, layer=0, protection_absent=1
+    *ptr++ = 0xf9;  // b11111001, ID=1(MPEG-2), layer=0, protection_absent=1
 
     *ptr++ =
         profile << 6
-- 
cgit v1.1


From f590f948e6c0fe3c54008dd7802fd513f127792d Mon Sep 17 00:00:00 2001
From: Eric Laurent <elaurent@google.com>
Date: Fri, 6 Dec 2013 11:51:42 -0800
Subject: stagefright: do not offload LD-AAC decoding

For now, do not offload LD and ELD AAC decoding because there
is no way to know if it is supported by the audio DSP implementation.
The longer term fix will be to have mapMimeToAudioFormat() use the
audio object type in track metadata to refine the AAC format and the
audio HAL list supported AAC profiles.

Bug: 11697128.

Change-Id: Iaa88ecf3f4ae42ad48c1b42a9b007dd80eb88147
---
 media/libstagefright/MPEG4Extractor.cpp |  3 +++
 media/libstagefright/Utils.cpp          | 11 +++++++++++
 2 files changed, 14 insertions(+)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 6f51c39..491b4d1 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -2297,6 +2297,9 @@ status_t MPEG4Extractor::updateAudioTrackInfoFromESDS_MPEG4Audio(
         objectType = 32 + br.getBits(6);
     }
 
+    //keep AOT type
+    mLastTrack->meta->setInt32(kKeyAACAOT, objectType);
+
     uint32_t freqIndex = br.getBits(4);
 
     int32_t sampleRate = 0;
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index 9041c21..216a329 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -562,6 +562,17 @@ bool canOffloadStream(const sp<MetaData>& meta, bool hasVideo,
         return false;
     }
 
+    // check whether it is ELD/LD content -> no offloading
+    // FIXME: this should depend on audio DSP capabilities. mapMimeToAudioFormat() should use the
+    // metadata to refine the AAC format and the audio HAL should only list supported profiles.
+    int32_t aacaot = -1;
+    if (meta->findInt32(kKeyAACAOT, &aacaot)) {
+        if (aacaot == 23 || aacaot == 39 ) {
+            ALOGV("track of type '%s' is ELD/LD content", mime);
+            return false;
+        }
+    }
+
     int32_t srate = -1;
     if (!meta->findInt32(kKeySampleRate, &srate)) {
         ALOGV("track of type '%s' does not publish sample rate", mime);
-- 
cgit v1.1


From 6708a45ba1b5eacf583d5e80b2075b68940fe0ab Mon Sep 17 00:00:00 2001
From: Lajos Molnar <lajos@google.com>
Date: Mon, 16 Dec 2013 10:16:32 -0800
Subject: stagefright: Fix bugs in playlist fetcher logic

1. Improve start time: start playback as soon as the target duration
or 10s was buffered.
2. Select playlist monitor time based on target duration to avoid
continuously missing the boat.
3. If "we miss the boat" we still must request a safe sequence
number (last - 3)

Change-Id: Ie99c360ac67b152ad9af19e9c6e520016f67e4e3
Signed-off-by: Lajos Molnar <lajos@google.com>
Bug: 12060952
---
 media/libstagefright/httplive/PlaylistFetcher.cpp | 127 +++++++++++++++++-----
 media/libstagefright/httplive/PlaylistFetcher.h   |   7 +-
 2 files changed, 105 insertions(+), 29 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index 973b779..1754bf2 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -47,6 +47,7 @@ namespace android {
 
 // static
 const int64_t PlaylistFetcher::kMinBufferedDurationUs = 10000000ll;
+const int64_t PlaylistFetcher::kMaxMonitorDelayUs = 3000000ll;
 
 PlaylistFetcher::PlaylistFetcher(
         const sp<AMessage> &notify,
@@ -61,6 +62,7 @@ PlaylistFetcher::PlaylistFetcher(
       mSeqNumber(-1),
       mNumRetries(0),
       mStartup(true),
+      mPrepared(false),
       mNextPTSTimeUs(-1ll),
       mMonitorQueueGeneration(0),
       mRefreshState(INITIAL_MINIMUM_RELOAD_DELAY),
@@ -103,10 +105,16 @@ int64_t PlaylistFetcher::getSegmentStartTimeUs(int32_t seqNumber) const {
     return segmentStartUs;
 }
 
-bool PlaylistFetcher::timeToRefreshPlaylist(int64_t nowUs) const {
-    if (mPlaylist == NULL) {
+int64_t PlaylistFetcher::delayUsToRefreshPlaylist() const {
+    int64_t nowUs = ALooper::GetNowUs();
+
+    if (mPlaylist == NULL || mLastPlaylistFetchTimeUs < 0ll) {
         CHECK_EQ((int)mRefreshState, (int)INITIAL_MINIMUM_RELOAD_DELAY);
-        return true;
+        return 0ll;
+    }
+
+    if (mPlaylist->isComplete()) {
+        return (~0llu >> 1);
     }
 
     int32_t targetDurationSecs;
@@ -157,7 +165,8 @@ bool PlaylistFetcher::timeToRefreshPlaylist(int64_t nowUs) const {
             break;
     }
 
-    return mLastPlaylistFetchTimeUs + minPlaylistAgeUs <= nowUs;
+    int64_t delayUs = mLastPlaylistFetchTimeUs + minPlaylistAgeUs - nowUs;
+    return delayUs > 0ll ? delayUs : 0ll;
 }
 
 status_t PlaylistFetcher::decryptBuffer(
@@ -274,7 +283,15 @@ status_t PlaylistFetcher::decryptBuffer(
     return OK;
 }
 
-void PlaylistFetcher::postMonitorQueue(int64_t delayUs) {
+void PlaylistFetcher::postMonitorQueue(int64_t delayUs, int64_t minDelayUs) {
+    int64_t maxDelayUs = delayUsToRefreshPlaylist();
+    if (maxDelayUs < minDelayUs) {
+        maxDelayUs = minDelayUs;
+    }
+    if (delayUs > maxDelayUs) {
+        ALOGV("Need to refresh playlist in %lld", maxDelayUs);
+        delayUs = maxDelayUs;
+    }
     sp<AMessage> msg = new AMessage(kWhatMonitorQueue, id());
     msg->setInt32("generation", mMonitorQueueGeneration);
     msg->post(delayUs);
@@ -415,6 +432,7 @@ status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) {
     if (mStartTimeUs >= 0ll) {
         mSeqNumber = -1;
         mStartup = true;
+        mPrepared = false;
     }
 
     postMonitorQueue();
@@ -456,40 +474,62 @@ void PlaylistFetcher::queueDiscontinuity(
 
 void PlaylistFetcher::onMonitorQueue() {
     bool downloadMore = false;
+    refreshPlaylist();
+
+    int32_t targetDurationSecs;
+    int64_t targetDurationUs = kMinBufferedDurationUs;
+    if (mPlaylist != NULL) {
+        CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs));
+        targetDurationUs = targetDurationSecs * 1000000ll;
+    }
 
-    status_t finalResult;
+    // buffer at least 3 times the target duration, or up to 10 seconds
+    int64_t durationToBufferUs = targetDurationUs * 3;
+    if (durationToBufferUs > kMinBufferedDurationUs)  {
+        durationToBufferUs = kMinBufferedDurationUs;
+    }
+
+    int64_t bufferedDurationUs = 0ll;
+    status_t finalResult = NOT_ENOUGH_DATA;
     if (mStreamTypeMask == LiveSession::STREAMTYPE_SUBTITLES) {
         sp<AnotherPacketSource> packetSource =
             mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES);
 
-        int64_t bufferedDurationUs =
+        bufferedDurationUs =
                 packetSource->getBufferedDurationUs(&finalResult);
-
-        downloadMore = (bufferedDurationUs < kMinBufferedDurationUs);
         finalResult = OK;
     } else {
         bool first = true;
-        int64_t minBufferedDurationUs = 0ll;
 
         for (size_t i = 0; i < mPacketSources.size(); ++i) {
             if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0) {
                 continue;
             }
 
-            int64_t bufferedDurationUs =
+            int64_t bufferedStreamDurationUs =
                 mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult);
-
-            if (first || bufferedDurationUs < minBufferedDurationUs) {
-                minBufferedDurationUs = bufferedDurationUs;
+            if (first || bufferedStreamDurationUs < bufferedDurationUs) {
+                bufferedDurationUs = bufferedStreamDurationUs;
                 first = false;
             }
         }
+    }
+    downloadMore = (bufferedDurationUs < durationToBufferUs);
+
+    // signal start if buffered up at least the target size
+    if (!mPrepared && bufferedDurationUs > targetDurationUs && downloadMore) {
+        mPrepared = true;
 
-        downloadMore =
-            !first && (minBufferedDurationUs < kMinBufferedDurationUs);
+        ALOGV("prepared, buffered=%lld > %lld",
+                bufferedDurationUs, targetDurationUs);
+        sp<AMessage> msg = mNotify->dup();
+        msg->setInt32("what", kWhatTemporarilyDoneFetching);
+        msg->post();
     }
 
     if (finalResult == OK && downloadMore) {
+        ALOGV("monitoring, buffered=%lld < %lld",
+                bufferedDurationUs, durationToBufferUs);
         onDownloadNext();
     } else {
         // Nothing to do yet, try again in a second.
@@ -498,15 +538,17 @@ void PlaylistFetcher::onMonitorQueue() {
         msg->setInt32("what", kWhatTemporarilyDoneFetching);
         msg->post();
 
-        postMonitorQueue(1000000ll);
+        int64_t delayUs = mPrepared ? kMaxMonitorDelayUs : targetDurationUs / 2;
+        ALOGV("pausing for %lld, buffered=%lld > %lld",
+                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);
     }
 }
 
-void PlaylistFetcher::onDownloadNext() {
-    int64_t nowUs = ALooper::GetNowUs();
-
-    if (mLastPlaylistFetchTimeUs < 0ll
-            || (!mPlaylist->isComplete() && timeToRefreshPlaylist(nowUs))) {
+status_t PlaylistFetcher::refreshPlaylist() {
+    if (delayUsToRefreshPlaylist() <= 0) {
         bool unchanged;
         sp<M3UParser> playlist = mSession->fetchPlaylist(
                 mURI.c_str(), mPlaylistHash, &unchanged);
@@ -522,7 +564,7 @@ void PlaylistFetcher::onDownloadNext() {
             } else {
                 ALOGE("failed to load playlist at url '%s'", mURI.c_str());
                 notifyError(ERROR_IO);
-                return;
+                return ERROR_IO;
             }
         } else {
             mRefreshState = INITIAL_MINIMUM_RELOAD_DELAY;
@@ -535,6 +577,13 @@ void PlaylistFetcher::onDownloadNext() {
 
         mLastPlaylistFetchTimeUs = ALooper::GetNowUs();
     }
+    return OK;
+}
+
+void PlaylistFetcher::onDownloadNext() {
+    if (refreshPlaylist() != OK) {
+        return;
+    }
 
     int32_t firstSeqNumberInPlaylist;
     if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32(
@@ -553,12 +602,18 @@ void PlaylistFetcher::onDownloadNext() {
 
         if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
             mSeqNumber = getSeqNumberForTime(mStartTimeUs);
+            ALOGV("Initial sequence number for time %lld is %ld from (%ld .. %ld)",
+                    mStartTimeUs, mSeqNumber, firstSeqNumberInPlaylist,
+                    lastSeqNumberInPlaylist);
         } else {
             // If this is a live session, start 3 segments from the end.
             mSeqNumber = lastSeqNumberInPlaylist - 3;
             if (mSeqNumber < firstSeqNumberInPlaylist) {
                 mSeqNumber = firstSeqNumberInPlaylist;
             }
+            ALOGV("Initial sequence number for live event %ld from (%ld .. %ld)",
+                    mSeqNumber, firstSeqNumberInPlaylist,
+                    lastSeqNumberInPlaylist);
         }
 
         mStartTimeUs = -1ll;
@@ -570,16 +625,34 @@ void PlaylistFetcher::onDownloadNext() {
             ++mNumRetries;
 
             if (mSeqNumber > lastSeqNumberInPlaylist) {
-                mLastPlaylistFetchTimeUs = -1;
-                postMonitorQueue(3000000ll);
+                // refresh in increasing fraction (1/2, 1/3, ...) of the
+                // playlist's target duration or 3 seconds, whichever is less
+                int32_t targetDurationSecs;
+                CHECK(mPlaylist->meta()->findInt32(
+                        "target-duration", &targetDurationSecs));
+                int64_t delayUs = mPlaylist->size() * targetDurationSecs *
+                        1000000ll / (1 + mNumRetries);
+                if (delayUs > kMaxMonitorDelayUs) {
+                    delayUs = kMaxMonitorDelayUs;
+                }
+                ALOGV("sequence number high: %ld from (%ld .. %ld), monitor in %lld (retry=%d)",
+                        mSeqNumber, firstSeqNumberInPlaylist,
+                        lastSeqNumberInPlaylist, delayUs, mNumRetries);
+                postMonitorQueue(delayUs);
                 return;
             }
 
             // we've missed the boat, let's start from the lowest sequence
             // number available and signal a discontinuity.
 
-            ALOGI("We've missed the boat, restarting playback.");
-            mSeqNumber = lastSeqNumberInPlaylist;
+            ALOGI("We've missed the boat, restarting playback."
+                  "  mStartup=%d, was  looking for %d in %d-%d",
+                    mStartup, mSeqNumber, firstSeqNumberInPlaylist,
+                    lastSeqNumberInPlaylist);
+            mSeqNumber = lastSeqNumberInPlaylist - 3;
+            if (mSeqNumber < firstSeqNumberInPlaylist) {
+                mSeqNumber = firstSeqNumberInPlaylist;
+            }
             explicitDiscontinuity = true;
 
             // fall through
diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h
index 1648e02..78dea20 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.h
+++ b/media/libstagefright/httplive/PlaylistFetcher.h
@@ -79,6 +79,7 @@ private:
     };
 
     static const int64_t kMinBufferedDurationUs;
+    static const int64_t kMaxMonitorDelayUs;
 
     sp<AMessage> mNotify;
     sp<LiveSession> mSession;
@@ -97,6 +98,7 @@ private:
     int32_t mSeqNumber;
     int32_t mNumRetries;
     bool mStartup;
+    bool mPrepared;
     int64_t mNextPTSTimeUs;
 
     int32_t mMonitorQueueGeneration;
@@ -120,10 +122,11 @@ private:
     status_t decryptBuffer(
             size_t playlistIndex, const sp<ABuffer> &buffer);
 
-    void postMonitorQueue(int64_t delayUs = 0);
+    void postMonitorQueue(int64_t delayUs = 0, int64_t minDelayUs = 0);
     void cancelMonitorQueue();
 
-    bool timeToRefreshPlaylist(int64_t nowUs) const;
+    int64_t delayUsToRefreshPlaylist() const;
+    status_t refreshPlaylist();
 
     // Returns the media time in us of the segment specified by seqNumber.
     // This is computed by summing the durations of all segments before it.
-- 
cgit v1.1


From e62d20268de87f63083896b5eef083d541078322 Mon Sep 17 00:00:00 2001
From: Lajos Molnar <lajos@google.com>
Date: Tue, 17 Dec 2013 14:10:46 -0800
Subject: stagefright: Fix issue with tracking media format in packet source

Media format in AnotherPacketSource is now tracked across discontinuities.
This fixes a bug where format was set on queueAccessUnit and cleared on
dequeueAccessUnit, thereby allowing it to remain cleared.

Change-Id: I20975a630443f4a223a2b4344e8244f34b9560b9
Signed-off-by: Lajos Molnar <lajos@google.com>
Bug: 12060952
---
 media/libstagefright/httplive/PlaylistFetcher.cpp  |  9 ++--
 .../libstagefright/mpeg2ts/AnotherPacketSource.cpp | 48 ++++++++++++++++++----
 2 files changed, 44 insertions(+), 13 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index 1754bf2..f095987 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -861,12 +861,13 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits(
                     && source->dequeueAccessUnit(&accessUnit) == OK) {
                 // Note that we do NOT dequeue any discontinuities.
 
+                // for simplicity, store a reference to the format in each unit
+                sp<MetaData> format = source->getFormat();
+                if (format != NULL) {
+                    accessUnit->meta()->setObject("format", format);
+                }
                 packetSource->queueAccessUnit(accessUnit);
             }
-
-            if (packetSource->getFormat() == NULL) {
-                packetSource->setFormat(source->getFormat());
-            }
         }
 
         return OK;
diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
index 3153c8b..52fb2a5 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
@@ -70,7 +70,27 @@ status_t AnotherPacketSource::stop() {
 }
 
 sp<MetaData> AnotherPacketSource::getFormat() {
-    return mFormat;
+    Mutex::Autolock autoLock(mLock);
+    if (mFormat != NULL) {
+        return mFormat;
+    }
+
+    List<sp<ABuffer> >::iterator it = mBuffers.begin();
+    while (it != mBuffers.end()) {
+        sp<ABuffer> buffer = *it;
+        int32_t discontinuity;
+        if (buffer->meta()->findInt32("discontinuity", &discontinuity)) {
+            break;
+        }
+
+        sp<RefBase> object;
+        if (buffer->meta()->findObject("format", &object)) {
+            return static_cast<MetaData*>(object.get());
+        }
+
+        ++it;
+    }
+    return NULL;
 }
 
 status_t AnotherPacketSource::dequeueAccessUnit(sp<ABuffer> *buffer) {
@@ -94,6 +114,11 @@ status_t AnotherPacketSource::dequeueAccessUnit(sp<ABuffer> *buffer) {
             return INFO_DISCONTINUITY;
         }
 
+        sp<RefBase> object;
+        if ((*buffer)->meta()->findObject("format", &object)) {
+            mFormat = static_cast<MetaData*>(object.get());
+        }
+
         return OK;
     }
 
@@ -120,17 +145,22 @@ status_t AnotherPacketSource::read(
             }
 
             return INFO_DISCONTINUITY;
-        } else {
-            int64_t timeUs;
-            CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
+        }
 
-            MediaBuffer *mediaBuffer = new MediaBuffer(buffer);
+        sp<RefBase> object;
+        if (buffer->meta()->findObject("format", &object)) {
+            mFormat = static_cast<MetaData*>(object.get());
+        }
 
-            mediaBuffer->meta_data()->setInt64(kKeyTime, timeUs);
+        int64_t timeUs;
+        CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
 
-            *out = mediaBuffer;
-            return OK;
-        }
+        MediaBuffer *mediaBuffer = new MediaBuffer(buffer);
+
+        mediaBuffer->meta_data()->setInt64(kKeyTime, timeUs);
+
+        *out = mediaBuffer;
+        return OK;
     }
 
     return mEOSResult;
-- 
cgit v1.1


From b2be2f2e41d5502d79636b21cbad23f4f983bbd7 Mon Sep 17 00:00:00 2001
From: Robert Shih <robertshih@google.com>
Date: Thu, 23 Jan 2014 14:16:32 -0800
Subject: Fixed misplaced bracket for switch-case block.

Bug: 11854054
Change-Id: I66b410e1e19601ad65d8357bda4c0c58e2dc15dc
---
 media/libstagefright/ACodec.cpp | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 5d5220f..bc4e89b 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -3247,11 +3247,11 @@ void ACodec::BaseState::onInputBufferFilled(const sp<AMessage> &msg) {
                 mCodec->mInputEOSResult = err;
             }
             break;
-
-            default:
-                CHECK_EQ((int)mode, (int)FREE_BUFFERS);
-                break;
         }
+
+        default:
+            CHECK_EQ((int)mode, (int)FREE_BUFFERS);
+            break;
     }
 }
 
-- 
cgit v1.1


From f7e34cfcb9d649989c8e391f23dde782d4931849 Mon Sep 17 00:00:00 2001
From: Robert Shih <robertshih@google.com>
Date: Thu, 23 Jan 2014 15:00:53 -0800
Subject: stagefright: use CODECS field in EXT-X_STREAM-INF as fallback

Use the CODECS field in EXT-X-STREAM-INF as a viable fallback for identifying
available streams in a variant playlist.

Bug: 11854054
Change-Id: I5011809dc7c56220b023eb1a16f2d6a392ddd6f1
---
 media/libstagefright/httplive/M3UParser.cpp | 160 ++++++++++++++++++++++++++--
 media/libstagefright/httplive/M3UParser.h   |   4 +
 2 files changed, 157 insertions(+), 7 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp
index 243888c..ae19ffa 100644
--- a/media/libstagefright/httplive/M3UParser.cpp
+++ b/media/libstagefright/httplive/M3UParser.cpp
@@ -24,6 +24,7 @@
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AMessage.h>
 #include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/Utils.h>
 #include <media/mediaplayer.h>
 
 namespace android {
@@ -352,9 +353,27 @@ bool M3UParser::getTypeURI(size_t index, const char *key, AString *uri) const {
     if (!meta->findString(key, &groupID)) {
         *uri = mItems.itemAt(index).mURI;
 
-        // Assume media without any more specific attribute contains
-        // audio and video, but no subtitles.
-        return !strcmp("audio", key) || !strcmp("video", key);
+        AString codecs;
+        if (!meta->findString("codecs", &codecs)) {
+            // Assume media without any more specific attribute contains
+            // audio and video, but no subtitles.
+            return !strcmp("audio", key) || !strcmp("video", key);
+        } else {
+            // Split the comma separated list of codecs.
+            size_t offset = 0;
+            ssize_t commaPos = -1;
+            codecs.append(',');
+            while ((commaPos = codecs.find(",", offset)) >= 0) {
+                AString codec(codecs, offset, commaPos - offset);
+                // return true only if a codec of type `key` ("audio"/"video")
+                // is found.
+                if (codecIsType(codec, key)) {
+                    return true;
+                }
+                offset = commaPos + 1;
+            }
+            return false;
+        }
     }
 
     sp<MediaGroup> group = mMediaGroups.valueFor(groupID);
@@ -684,12 +703,22 @@ status_t M3UParser::parseStreamInf(
                 *meta = new AMessage;
             }
             (*meta)->setInt32("bandwidth", x);
+        } else if (!strcasecmp("codecs", key.c_str())) {
+            if (!isQuotedString(val)) {
+                ALOGE("Expected quoted string for %s attribute, "
+                      "got '%s' instead.",
+                      key.c_str(), val.c_str());;
+
+                return ERROR_MALFORMED;
+            }
+
+            key.tolower();
+            const AString &codecs = unquoteString(val);
+            (*meta)->setString(key.c_str(), codecs.c_str());
         } else if (!strcasecmp("audio", key.c_str())
                 || !strcasecmp("video", key.c_str())
                 || !strcasecmp("subtitles", key.c_str())) {
-            if (val.size() < 2
-                    || val.c_str()[0] != '"'
-                    || val.c_str()[val.size() - 1] != '"') {
+            if (!isQuotedString(val)) {
                 ALOGE("Expected quoted string for %s attribute, "
                       "got '%s' instead.",
                       key.c_str(), val.c_str());
@@ -697,7 +726,7 @@ status_t M3UParser::parseStreamInf(
                 return ERROR_MALFORMED;
             }
 
-            AString groupID(val, 1, val.size() - 2);
+            const AString &groupID = unquoteString(val);
             ssize_t groupIndex = mMediaGroups.indexOfKey(groupID);
 
             if (groupIndex < 0) {
@@ -1086,4 +1115,121 @@ status_t M3UParser::ParseDouble(const char *s, double *x) {
     return OK;
 }
 
+// static
+bool M3UParser::isQuotedString(const AString &str) {
+    if (str.size() < 2
+            || str.c_str()[0] != '"'
+            || str.c_str()[str.size() - 1] != '"') {
+        return false;
+    }
+    return true;
+}
+
+// static
+AString M3UParser::unquoteString(const AString &str) {
+     if (!isQuotedString(str)) {
+         return str;
+     }
+     return AString(str, 1, str.size() - 2);
+}
+
+// static
+bool M3UParser::codecIsType(const AString &codec, const char *type) {
+    if (codec.size() < 4) {
+        return false;
+    }
+    const char *c = codec.c_str();
+    switch (FOURCC(c[0], c[1], c[2], c[3])) {
+        // List extracted from http://www.mp4ra.org/codecs.html
+        case 'ac-3':
+        case 'alac':
+        case 'dra1':
+        case 'dtsc':
+        case 'dtse':
+        case 'dtsh':
+        case 'dtsl':
+        case 'ec-3':
+        case 'enca':
+        case 'g719':
+        case 'g726':
+        case 'm4ae':
+        case 'mlpa':
+        case 'mp4a':
+        case 'raw ':
+        case 'samr':
+        case 'sawb':
+        case 'sawp':
+        case 'sevc':
+        case 'sqcp':
+        case 'ssmv':
+        case 'twos':
+        case 'agsm':
+        case 'alaw':
+        case 'dvi ':
+        case 'fl32':
+        case 'fl64':
+        case 'ima4':
+        case 'in24':
+        case 'in32':
+        case 'lpcm':
+        case 'Qclp':
+        case 'QDM2':
+        case 'QDMC':
+        case 'ulaw':
+        case 'vdva':
+            return !strcmp("audio", type);
+
+        case 'avc1':
+        case 'avc2':
+        case 'avcp':
+        case 'drac':
+        case 'encv':
+        case 'mjp2':
+        case 'mp4v':
+        case 'mvc1':
+        case 'mvc2':
+        case 'resv':
+        case 's263':
+        case 'svc1':
+        case 'vc-1':
+        case 'CFHD':
+        case 'civd':
+        case 'DV10':
+        case 'dvh5':
+        case 'dvh6':
+        case 'dvhp':
+        case 'DVOO':
+        case 'DVOR':
+        case 'DVTV':
+        case 'DVVT':
+        case 'flic':
+        case 'gif ':
+        case 'h261':
+        case 'h263':
+        case 'HD10':
+        case 'jpeg':
+        case 'M105':
+        case 'mjpa':
+        case 'mjpb':
+        case 'png ':
+        case 'PNTG':
+        case 'rle ':
+        case 'rpza':
+        case 'Shr0':
+        case 'Shr1':
+        case 'Shr2':
+        case 'Shr3':
+        case 'Shr4':
+        case 'SVQ1':
+        case 'SVQ3':
+        case 'tga ':
+        case 'tiff':
+        case 'WRLE':
+            return !strcmp("video", type);
+
+        default:
+            return false;
+    }
+}
+
 }  // namespace android
diff --git a/media/libstagefright/httplive/M3UParser.h b/media/libstagefright/httplive/M3UParser.h
index 5248004..b93b0e5 100644
--- a/media/libstagefright/httplive/M3UParser.h
+++ b/media/libstagefright/httplive/M3UParser.h
@@ -100,6 +100,10 @@ private:
     static status_t ParseInt32(const char *s, int32_t *x);
     static status_t ParseDouble(const char *s, double *x);
 
+    static bool isQuotedString(const AString &str);
+    static AString unquoteString(const AString &str);
+    static bool codecIsType(const AString &codec, const char *type);
+
     DISALLOW_EVIL_CONSTRUCTORS(M3UParser);
 };
 
-- 
cgit v1.1


From 933a4d3339ebbcd34a7f97b9e7350ec74b5ec29c Mon Sep 17 00:00:00 2001
From: Robert Shih <robertshih@google.com>
Date: Thu, 23 Jan 2014 15:26:43 -0800
Subject: Added support to query ACodec whether adaptive playback is enabled.

Bug: 11854054
Change-Id: I6b0308aa8550c643706959277e46dad586c37297
---
 media/libstagefright/ACodec.cpp | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index bc4e89b..3810ac1 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -363,6 +363,7 @@ ACodec::ACodec()
       mIsEncoder(false),
       mUseMetadataOnEncoderOutput(false),
       mShutdownInProgress(false),
+      mIsConfiguredForAdaptivePlayback(false),
       mEncoderDelay(0),
       mEncoderPadding(0),
       mChannelMaskPresent(false),
@@ -1115,6 +1116,7 @@ status_t ACodec::configureCodec(
     int32_t haveNativeWindow = msg->findObject("native-window", &obj) &&
             obj != NULL;
     mStoreMetaDataInOutputBuffers = false;
+    mIsConfiguredForAdaptivePlayback = false;
     if (!encoder && video && haveNativeWindow) {
         err = mOMX->storeMetaDataInBuffers(mNode, kPortIndexOutput, OMX_TRUE);
         if (err != OK) {
@@ -1159,12 +1161,14 @@ status_t ACodec::configureCodec(
                 ALOGW_IF(err != OK,
                         "[%s] prepareForAdaptivePlayback failed w/ err %d",
                         mComponentName.c_str(), err);
+                mIsConfiguredForAdaptivePlayback = (err == OK);
             }
             // allow failure
             err = OK;
         } else {
             ALOGV("[%s] storeMetaDataInBuffers succeeded", mComponentName.c_str());
             mStoreMetaDataInOutputBuffers = true;
+            mIsConfiguredForAdaptivePlayback = true;
         }
 
         int32_t push;
@@ -3702,6 +3706,7 @@ void ACodec::LoadedState::stateEntered() {
     mCodec->mDequeueCounter = 0;
     mCodec->mMetaDataBuffersToSubmit = 0;
     mCodec->mRepeatFrameDelayUs = -1ll;
+    mCodec->mIsConfiguredForAdaptivePlayback = false;
 
     if (mCodec->mShutdownInProgress) {
         bool keepComponentAllocated = mCodec->mKeepComponentAllocated;
-- 
cgit v1.1


From 7e50e1c0c10cba1e27cafe581273adcadf93877d Mon Sep 17 00:00:00 2001
From: Robert Shih <robertshih@google.com>
Date: Thu, 23 Jan 2014 14:25:32 -0800
Subject: AnotherPacketSource support to get latest buffered MetaData.

Bug: 11854054
Change-Id: Ib3b6e0984036082bf3c4eb7901a2b29be52fdd29
---
 .../libstagefright/mpeg2ts/AnotherPacketSource.cpp | 24 ++++++++++++++++++++--
 media/libstagefright/mpeg2ts/AnotherPacketSource.h |  3 +++
 2 files changed, 25 insertions(+), 2 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
index 52fb2a5..2b0bf30 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
@@ -34,7 +34,8 @@ AnotherPacketSource::AnotherPacketSource(const sp<MetaData> &meta)
     : mIsAudio(false),
       mFormat(NULL),
       mLastQueuedTimeUs(0),
-      mEOSResult(OK) {
+      mEOSResult(OK),
+      mLatestEnqueuedMeta(NULL) {
     setFormat(meta);
 }
 
@@ -182,12 +183,24 @@ void AnotherPacketSource::queueAccessUnit(const sp<ABuffer> &buffer) {
         return;
     }
 
-    CHECK(buffer->meta()->findInt64("timeUs", &mLastQueuedTimeUs));
+    int64_t lastQueuedTimeUs;
+    CHECK(buffer->meta()->findInt64("timeUs", &lastQueuedTimeUs));
+    mLastQueuedTimeUs = lastQueuedTimeUs;
     ALOGV("queueAccessUnit timeUs=%lld us (%.2f secs)", mLastQueuedTimeUs, mLastQueuedTimeUs / 1E6);
 
     Mutex::Autolock autoLock(mLock);
     mBuffers.push_back(buffer);
     mCondition.signal();
+
+    if (!mLatestEnqueuedMeta.get()) {
+        mLatestEnqueuedMeta = buffer->meta();
+    } else {
+        int64_t latestTimeUs = 0;
+        CHECK(mLatestEnqueuedMeta->findInt64("timeUs", &latestTimeUs));
+        if (lastQueuedTimeUs > latestTimeUs) {
+            mLatestEnqueuedMeta = buffer->meta();
+        }
+    }
 }
 
 void AnotherPacketSource::clear() {
@@ -197,6 +210,7 @@ void AnotherPacketSource::clear() {
     mEOSResult = OK;
 
     mFormat = NULL;
+    mLatestEnqueuedMeta = NULL;
 }
 
 void AnotherPacketSource::queueDiscontinuity(
@@ -221,6 +235,7 @@ void AnotherPacketSource::queueDiscontinuity(
 
     mEOSResult = OK;
     mLastQueuedTimeUs = 0;
+    mLatestEnqueuedMeta = NULL;
 
     sp<ABuffer> buffer = new ABuffer(0);
     buffer->meta()->setInt32("discontinuity", static_cast<int32_t>(type));
@@ -308,4 +323,9 @@ bool AnotherPacketSource::isFinished(int64_t duration) const {
     return (mEOSResult != OK);
 }
 
+sp<AMessage> AnotherPacketSource::getLatestMeta() {
+    Mutex::Autolock autoLock(mLock);
+    return mLatestEnqueuedMeta;
+}
+
 }  // namespace android
diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.h b/media/libstagefright/mpeg2ts/AnotherPacketSource.h
index e16cf78..9b193a2 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.h
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.h
@@ -62,6 +62,8 @@ struct AnotherPacketSource : public MediaSource {
 
     bool isFinished(int64_t duration) const;
 
+    sp<AMessage> getLatestMeta();
+
 protected:
     virtual ~AnotherPacketSource();
 
@@ -74,6 +76,7 @@ private:
     int64_t mLastQueuedTimeUs;
     List<sp<ABuffer> > mBuffers;
     status_t mEOSResult;
+    sp<AMessage> mLatestEnqueuedMeta;
 
     bool wasFormatChange(int32_t discontinuityType) const;
 
-- 
cgit v1.1


From 822a489e595336be447f47f5c2a051e8fdd1cdff Mon Sep 17 00:00:00 2001
From: Robert Shih <robertshih@google.com>
Date: Thu, 6 Feb 2014 14:01:30 -0800
Subject: LiveSession: Add support for block-by-block fetchFile.

Bug: 11854054
Change-Id: I4025ba7fab8fab2e0c720f73894e908fd98a43d8
---
 media/libstagefright/httplive/LiveSession.cpp | 97 +++++++++++++++++----------
 media/libstagefright/httplive/LiveSession.h   | 18 ++++-
 2 files changed, 80 insertions(+), 35 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index bd12ddc..0333a2a 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -512,53 +512,80 @@ sp<PlaylistFetcher> LiveSession::addFetcher(const char *uri) {
     return info.mFetcher;
 }
 
+/*
+ * Illustration of parameters:
+ *
+ * 0      `range_offset`
+ * +------------+-------------------------------------------------------+--+--+
+ * |            |                                 | next block to fetch |  |  |
+ * |            | `source` handle => `out` buffer |                     |  |  |
+ * | `url` file |<--------- buffer size --------->|<--- `block_size` -->|  |  |
+ * |            |<----------- `range_length` / buffer capacity ----------->|  |
+ * |<------------------------------ file_size ------------------------------->|
+ *
+ * Special parameter values:
+ * - range_length == -1 means entire file
+ * - block_size == 0 means entire range
+ *
+ */
 status_t LiveSession::fetchFile(
         const char *url, sp<ABuffer> *out,
-        int64_t range_offset, int64_t range_length) {
-    *out = NULL;
+        int64_t range_offset, int64_t range_length,
+        uint32_t block_size, /* download block size */
+        sp<DataSource> *source /* to return and reuse source */) {
+    off64_t size;
+    sp<DataSource> temp_source;
+    if (source == NULL) {
+        source = &temp_source;
+    }
 
-    sp<DataSource> source;
+    if (*source == NULL) {
+        if (!strncasecmp(url, "file://", 7)) {
+            *source = new FileSource(url + 7);
+        } else if (strncasecmp(url, "http://", 7)
+                && strncasecmp(url, "https://", 8)) {
+            return ERROR_UNSUPPORTED;
+        } else {
+            KeyedVector<String8, String8> headers = mExtraHeaders;
+            if (range_offset > 0 || range_length >= 0) {
+                headers.add(
+                        String8("Range"),
+                        String8(
+                            StringPrintf(
+                                "bytes=%lld-%s",
+                                range_offset,
+                                range_length < 0
+                                    ? "" : StringPrintf("%lld",
+                                            range_offset + range_length - 1).c_str()).c_str()));
+            }
+            status_t err = mHTTPDataSource->connect(url, &headers);
 
-    if (!strncasecmp(url, "file://", 7)) {
-        source = new FileSource(url + 7);
-    } else if (strncasecmp(url, "http://", 7)
-            && strncasecmp(url, "https://", 8)) {
-        return ERROR_UNSUPPORTED;
-    } else {
-        KeyedVector<String8, String8> headers = mExtraHeaders;
-        if (range_offset > 0 || range_length >= 0) {
-            headers.add(
-                    String8("Range"),
-                    String8(
-                        StringPrintf(
-                            "bytes=%lld-%s",
-                            range_offset,
-                            range_length < 0
-                                ? "" : StringPrintf("%lld", range_offset + range_length - 1).c_str()).c_str()));
-        }
-        status_t err = mHTTPDataSource->connect(url, &headers);
+            if (err != OK) {
+                return err;
+            }
 
-        if (err != OK) {
-            return err;
+            *source = mHTTPDataSource;
         }
-
-        source = mHTTPDataSource;
     }
 
-    off64_t size;
-    status_t err = source->getSize(&size);
-
-    if (err != OK) {
+    status_t getSizeErr = (*source)->getSize(&size);
+    if (getSizeErr != OK) {
         size = 65536;
     }
 
-    sp<ABuffer> buffer = new ABuffer(size);
-    buffer->setRange(0, 0);
+    sp<ABuffer> buffer = *out != NULL ? *out : new ABuffer(size);
+    if (*out == NULL) {
+        buffer->setRange(0, 0);
+    }
 
+    // adjust range_length if only reading partial block
+    if (block_size > 0 && (range_length == -1 || buffer->size() + block_size < range_length)) {
+        range_length = buffer->size() + block_size;
+    }
     for (;;) {
+        // Only resize when we don't know the size.
         size_t bufferRemaining = buffer->capacity() - buffer->size();
-
-        if (bufferRemaining == 0) {
+        if (bufferRemaining == 0 && getSizeErr != OK) {
             bufferRemaining = 32768;
 
             ALOGV("increasing download buffer to %d bytes",
@@ -583,7 +610,9 @@ status_t LiveSession::fetchFile(
             }
         }
 
-        ssize_t n = source->readAt(
+        // The DataSource is responsible for informing us of error (n < 0) or eof (n == 0)
+        // to help us break out of the loop.
+        ssize_t n = (*source)->readAt(
                 buffer->size(), buffer->data() + buffer->size(),
                 maxBytesToRead);
 
diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h
index 99b480a8..f42d7ae 100644
--- a/media/libstagefright/httplive/LiveSession.h
+++ b/media/libstagefright/httplive/LiveSession.h
@@ -145,9 +145,25 @@ private:
     status_t onSeek(const sp<AMessage> &msg);
     void onFinishDisconnect2();
 
+    // If given a non-zero block_size (default 0), it is used to cap the number of
+    // bytes read in from the DataSource. If given a non-NULL buffer, new content
+    // is read into the end.
+    //
+    // The DataSource we read from is responsible for signaling error or EOF to help us
+    // break out of the read loop. The DataSource can be returned to the caller, so
+    // that the caller can reuse it for subsequent fetches (within the initially
+    // requested range).
+    //
+    // For reused HTTP sources, the caller must download a file sequentially without
+    // any overlaps or gaps to prevent reconnection.
     status_t fetchFile(
             const char *url, sp<ABuffer> *out,
-            int64_t range_offset = 0, int64_t range_length = -1);
+            /* request/open a file starting at range_offset for range_length bytes */
+            int64_t range_offset = 0, int64_t range_length = -1,
+            /* download block size */
+            uint32_t block_size = 0,
+            /* reuse DataSource if doing partial fetch */
+            sp<DataSource> *source = NULL);
 
     sp<M3UParser> fetchPlaylist(
             const char *url, uint8_t *curPlaylistHash, bool *unchanged);
-- 
cgit v1.1


From 2cd94583868b775a548233a4f7cd1d988fc6344f Mon Sep 17 00:00:00 2001
From: Robert Shih <robertshih@google.com>
Date: Thu, 6 Feb 2014 14:25:25 -0800
Subject: PlaylistFetcher: Add support for block-by-block decryption.

Bug: 11854054
Change-Id: Ifd3f3369275889e716b360087b5b60d01635b578
---
 media/libstagefright/httplive/PlaylistFetcher.cpp | 95 ++++++++++++++---------
 media/libstagefright/httplive/PlaylistFetcher.h   | 17 +++-
 2 files changed, 75 insertions(+), 37 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index f095987..9cb16fe 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -170,7 +170,8 @@ int64_t PlaylistFetcher::delayUsToRefreshPlaylist() const {
 }
 
 status_t PlaylistFetcher::decryptBuffer(
-        size_t playlistIndex, const sp<ABuffer> &buffer) {
+        size_t playlistIndex, const sp<ABuffer> &buffer,
+        bool first) {
     sp<AMessage> itemMeta;
     bool found = false;
     AString method;
@@ -188,6 +189,7 @@ status_t PlaylistFetcher::decryptBuffer(
     if (!found) {
         method = "NONE";
     }
+    buffer->meta()->setString("cipher-method", method.c_str());
 
     if (method == "NONE") {
         return OK;
@@ -227,59 +229,77 @@ status_t PlaylistFetcher::decryptBuffer(
         return UNKNOWN_ERROR;
     }
 
-    unsigned char aes_ivec[16];
+    size_t n = buffer->size();
+    if (!n) {
+        return OK;
+    }
+    CHECK(n % 16 == 0);
 
-    AString iv;
-    if (itemMeta->findString("cipher-iv", &iv)) {
-        if ((!iv.startsWith("0x") && !iv.startsWith("0X"))
-                || iv.size() != 16 * 2 + 2) {
-            ALOGE("malformed cipher IV '%s'.", iv.c_str());
-            return ERROR_MALFORMED;
-        }
+    if (first) {
+        // If decrypting the first block in a file, read the iv from the manifest
+        // or derive the iv from the file's sequence number.
 
-        memset(aes_ivec, 0, sizeof(aes_ivec));
-        for (size_t i = 0; i < 16; ++i) {
-            char c1 = tolower(iv.c_str()[2 + 2 * i]);
-            char c2 = tolower(iv.c_str()[3 + 2 * i]);
-            if (!isxdigit(c1) || !isxdigit(c2)) {
+        AString iv;
+        if (itemMeta->findString("cipher-iv", &iv)) {
+            if ((!iv.startsWith("0x") && !iv.startsWith("0X"))
+                    || iv.size() != 16 * 2 + 2) {
                 ALOGE("malformed cipher IV '%s'.", iv.c_str());
                 return ERROR_MALFORMED;
             }
-            uint8_t nibble1 = isdigit(c1) ? c1 - '0' : c1 - 'a' + 10;
-            uint8_t nibble2 = isdigit(c2) ? c2 - '0' : c2 - 'a' + 10;
 
-            aes_ivec[i] = nibble1 << 4 | nibble2;
+            memset(mAESInitVec, 0, sizeof(mAESInitVec));
+            for (size_t i = 0; i < 16; ++i) {
+                char c1 = tolower(iv.c_str()[2 + 2 * i]);
+                char c2 = tolower(iv.c_str()[3 + 2 * i]);
+                if (!isxdigit(c1) || !isxdigit(c2)) {
+                    ALOGE("malformed cipher IV '%s'.", iv.c_str());
+                    return ERROR_MALFORMED;
+                }
+                uint8_t nibble1 = isdigit(c1) ? c1 - '0' : c1 - 'a' + 10;
+                uint8_t nibble2 = isdigit(c2) ? c2 - '0' : c2 - 'a' + 10;
+
+                mAESInitVec[i] = nibble1 << 4 | nibble2;
+            }
+        } else {
+            memset(mAESInitVec, 0, sizeof(mAESInitVec));
+            mAESInitVec[15] = mSeqNumber & 0xff;
+            mAESInitVec[14] = (mSeqNumber >> 8) & 0xff;
+            mAESInitVec[13] = (mSeqNumber >> 16) & 0xff;
+            mAESInitVec[12] = (mSeqNumber >> 24) & 0xff;
         }
-    } else {
-        memset(aes_ivec, 0, sizeof(aes_ivec));
-        aes_ivec[15] = mSeqNumber & 0xff;
-        aes_ivec[14] = (mSeqNumber >> 8) & 0xff;
-        aes_ivec[13] = (mSeqNumber >> 16) & 0xff;
-        aes_ivec[12] = (mSeqNumber >> 24) & 0xff;
     }
 
     AES_cbc_encrypt(
             buffer->data(), buffer->data(), buffer->size(),
-            &aes_key, aes_ivec, AES_DECRYPT);
-
-    // hexdump(buffer->data(), buffer->size());
+            &aes_key, mAESInitVec, AES_DECRYPT);
 
-    size_t n = buffer->size();
-    CHECK_GT(n, 0u);
+    return OK;
+}
 
-    size_t pad = buffer->data()[n - 1];
+status_t PlaylistFetcher::checkDecryptPadding(const sp<ABuffer> &buffer) {
+    status_t err;
+    AString method;
+    CHECK(buffer->meta()->findString("cipher-method", &method));
+    if (method == "NONE") {
+        return OK;
+    }
 
-    CHECK_GT(pad, 0u);
-    CHECK_LE(pad, 16u);
-    CHECK_GE((size_t)n, pad);
-    for (size_t i = 0; i < pad; ++i) {
-        CHECK_EQ((unsigned)buffer->data()[n - 1 - i], pad);
+    uint8_t padding = 0;
+    if (buffer->size() > 0) {
+        padding = buffer->data()[buffer->size() - 1];
     }
 
-    n -= pad;
+    if (padding > 16) {
+        return ERROR_MALFORMED;
+    }
 
-    buffer->setRange(buffer->offset(), n);
+    for (size_t i = buffer->size() - padding; i < padding; i++) {
+        if (buffer->data()[i] != padding) {
+            return ERROR_MALFORMED;
+        }
+    }
 
+    buffer->setRange(buffer->offset(), buffer->size() - padding);
     return OK;
 }
 
@@ -706,6 +726,9 @@ void PlaylistFetcher::onDownloadNext() {
     CHECK(buffer != NULL);
 
     err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, buffer);
+    if (err == OK) {
+        err = checkDecryptPadding(buffer);
+    }
 
     if (err != OK) {
         ALOGE("decryptBuffer failed w/ error %d", err);
diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h
index 78dea20..ac04a77 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.h
+++ b/media/libstagefright/httplive/PlaylistFetcher.h
@@ -119,8 +119,23 @@ private:
     uint64_t mFirstPTS;
     int64_t mAbsoluteTimeAnchorUs;
 
+    // Stores the initialization vector to decrypt the next block of cipher text, which can
+    // either be derived from the sequence number, read from the manifest, or copied from
+    // the last block of cipher text (cipher-block chaining).
+    unsigned char mAESInitVec[16];
+
+    // Set first to true if decrypting the first segment of a playlist segment. When
+    // first is true, reset the initialization vector based on the available
+    // information in the manifest; otherwise, use the initialization vector as
+    // updated by the last call to AES_cbc_encrypt.
+    //
+    // For the input to decrypt correctly, decryptBuffer must be called on
+    // consecutive byte ranges on block boundaries, e.g. 0..15, 16..47, 48..63,
+    // and so on.
     status_t decryptBuffer(
-            size_t playlistIndex, const sp<ABuffer> &buffer);
+            size_t playlistIndex, const sp<ABuffer> &buffer,
+            bool first = true);
+    status_t checkDecryptPadding(const sp<ABuffer> &buffer);
 
     void postMonitorQueue(int64_t delayUs = 0, int64_t minDelayUs = 0);
     void cancelMonitorQueue();
-- 
cgit v1.1


From 6c8495c8f1ccc35db972ee7ac0dbb8baf5843548 Mon Sep 17 00:00:00 2001
From: Robert Shih <robertshih@google.com>
Date: Fri, 14 Feb 2014 12:21:50 -0800
Subject: LiveSession refactor

Bug: 11854054
Change-Id: Ia91dce109835e042f72934376d4838b4cc72cb10
---
 media/libstagefright/httplive/LiveSession.cpp | 186 ++++++++++----------------
 media/libstagefright/httplive/LiveSession.h   |  28 +++-
 media/libstagefright/httplive/M3UParser.cpp   |  12 --
 media/libstagefright/httplive/M3UParser.h     |   6 +-
 4 files changed, 92 insertions(+), 140 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index 0333a2a..79ad100 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -66,14 +66,13 @@ LiveSession::LiveSession(
         mHTTPDataSource->setUID(mUID);
     }
 
-    mPacketSources.add(
-            STREAMTYPE_AUDIO, new AnotherPacketSource(NULL /* meta */));
+    mStreams[kAudioIndex] = StreamItem("audio");
+    mStreams[kVideoIndex] = StreamItem("video");
+    mStreams[kSubtitleIndex] = StreamItem("subtitle");
 
-    mPacketSources.add(
-            STREAMTYPE_VIDEO, new AnotherPacketSource(NULL /* meta */));
-
-    mPacketSources.add(
-            STREAMTYPE_SUBTITLES, new AnotherPacketSource(NULL /* meta */));
+    for (size_t i = 0; i < kMaxStreams; ++i) {
+        mPacketSources.add(indexToType(i), new AnotherPacketSource(NULL /* meta */));
+    }
 }
 
 LiveSession::~LiveSession() {
@@ -374,6 +373,12 @@ int LiveSession::SortByBandwidth(const BandwidthItem *a, const BandwidthItem *b)
     return 1;
 }
 
+// static
+LiveSession::StreamType LiveSession::indexToType(int idx) {
+    CHECK(idx >= 0 && idx < kMaxStreams);
+    return (StreamType)(1 << idx);
+}
+
 void LiveSession::onConnect(const sp<AMessage> &msg) {
     AString url;
     CHECK(msg->findString("url", &url));
@@ -858,19 +863,11 @@ void LiveSession::changeConfiguration(
 
     uint32_t streamMask = 0;
 
-    AString audioURI;
-    if (mPlaylist->getAudioURI(item.mPlaylistIndex, &audioURI)) {
-        streamMask |= STREAMTYPE_AUDIO;
-    }
-
-    AString videoURI;
-    if (mPlaylist->getVideoURI(item.mPlaylistIndex, &videoURI)) {
-        streamMask |= STREAMTYPE_VIDEO;
-    }
-
-    AString subtitleURI;
-    if (mPlaylist->getSubtitleURI(item.mPlaylistIndex, &subtitleURI)) {
-        streamMask |= STREAMTYPE_SUBTITLES;
+    AString URIs[kMaxStreams];
+    for (size_t i = 0; i < kMaxStreams; ++i) {
+        if (mPlaylist->getTypeURI(item.mPlaylistIndex, mStreams[i].mType, &URIs[i])) {
+            streamMask |= indexToType(i);
+        }
     }
 
     // Step 1, stop and discard fetchers that are no longer needed.
@@ -882,10 +879,10 @@ void LiveSession::changeConfiguration(
 
         // If we're seeking all current fetchers are discarded.
         if (timeUs < 0ll) {
-            if (((streamMask & STREAMTYPE_AUDIO) && uri == audioURI)
-                    || ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI)
-                    || ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI)) {
-                discardFetcher = false;
+            for (size_t j = 0; j < kMaxStreams; ++j) {
+                if ((streamMask & indexToType(j)) && uri == URIs[j]) {
+                    discardFetcher = false;
+                }
             }
         }
 
@@ -899,14 +896,10 @@ void LiveSession::changeConfiguration(
     sp<AMessage> msg = new AMessage(kWhatChangeConfiguration2, id());
     msg->setInt32("streamMask", streamMask);
     msg->setInt64("timeUs", timeUs);
-    if (streamMask & STREAMTYPE_AUDIO) {
-        msg->setString("audioURI", audioURI.c_str());
-    }
-    if (streamMask & STREAMTYPE_VIDEO) {
-        msg->setString("videoURI", videoURI.c_str());
-    }
-    if (streamMask & STREAMTYPE_SUBTITLES) {
-        msg->setString("subtitleURI", subtitleURI.c_str());
+    for (size_t i = 0; i < kMaxStreams; ++i) {
+        if (streamMask & indexToType(i)) {
+            msg->setString(mStreams[i].uriKey().c_str(), URIs[i].c_str());
+        }
     }
 
     // Every time a fetcher acknowledges the stopAsync or pauseAsync request
@@ -937,18 +930,13 @@ void LiveSession::onChangeConfiguration2(const sp<AMessage> &msg) {
     uint32_t streamMask;
     CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask));
 
-    AString audioURI, videoURI, subtitleURI;
-    if (streamMask & STREAMTYPE_AUDIO) {
-        CHECK(msg->findString("audioURI", &audioURI));
-        ALOGV("audioURI = '%s'", audioURI.c_str());
-    }
-    if (streamMask & STREAMTYPE_VIDEO) {
-        CHECK(msg->findString("videoURI", &videoURI));
-        ALOGV("videoURI = '%s'", videoURI.c_str());
-    }
-    if (streamMask & STREAMTYPE_SUBTITLES) {
-        CHECK(msg->findString("subtitleURI", &subtitleURI));
-        ALOGV("subtitleURI = '%s'", subtitleURI.c_str());
+    AString URIs[kMaxStreams];
+    for (size_t i = 0; i < kMaxStreams; ++i) {
+        if (streamMask & indexToType(i)) {
+            const AString &uriKey = mStreams[i].uriKey();
+            CHECK(msg->findString(uriKey.c_str(), &URIs[i]));
+            ALOGV("%s = '%s'", uriKey.c_str(), URIs[i].c_str());
+        }
     }
 
     // Determine which decoders to shutdown on the player side,
@@ -958,15 +946,12 @@ void LiveSession::onChangeConfiguration2(const sp<AMessage> &msg) {
     // 2) its streamtype was already active and still is but the URI
     //    has changed.
     uint32_t changedMask = 0;
-    if (((mStreamMask & streamMask & STREAMTYPE_AUDIO)
-                && !(audioURI == mAudioURI))
-        || (mStreamMask & ~streamMask & STREAMTYPE_AUDIO)) {
-        changedMask |= STREAMTYPE_AUDIO;
-    }
-    if (((mStreamMask & streamMask & STREAMTYPE_VIDEO)
-                && !(videoURI == mVideoURI))
-        || (mStreamMask & ~streamMask & STREAMTYPE_VIDEO)) {
-        changedMask |= STREAMTYPE_VIDEO;
+    for (size_t i = 0; i < kMaxStreams && i != kSubtitleIndex; ++i) {
+        if (((mStreamMask & streamMask & indexToType(i))
+                && !(URIs[i] == mStreams[i].mUri))
+                || (mStreamMask & ~streamMask & indexToType(i))) {
+            changedMask |= indexToType(i);
+        }
     }
 
     if (changedMask == 0) {
@@ -998,15 +983,10 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
     uint32_t streamMask;
     CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask));
 
-    AString audioURI, videoURI, subtitleURI;
-    if (streamMask & STREAMTYPE_AUDIO) {
-        CHECK(msg->findString("audioURI", &audioURI));
-    }
-    if (streamMask & STREAMTYPE_VIDEO) {
-        CHECK(msg->findString("videoURI", &videoURI));
-    }
-    if (streamMask & STREAMTYPE_SUBTITLES) {
-        CHECK(msg->findString("subtitleURI", &subtitleURI));
+    for (size_t i = 0; i < kMaxStreams; ++i) {
+        if (streamMask & indexToType(i)) {
+            CHECK(msg->findString(mStreams[i].uriKey().c_str(), &mStreams[i].mUri));
+        }
     }
 
     int64_t timeUs;
@@ -1018,9 +998,6 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
     mRealTimeBaseUs = ALooper::GetNowUs() - timeUs;
 
     mStreamMask = streamMask;
-    mAudioURI = audioURI;
-    mVideoURI = videoURI;
-    mSubtitleURI = subtitleURI;
 
     // Resume all existing fetchers and assign them packet sources.
     for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
@@ -1028,22 +1005,12 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
 
         uint32_t resumeMask = 0;
 
-        sp<AnotherPacketSource> audioSource;
-        if ((streamMask & STREAMTYPE_AUDIO) && uri == audioURI) {
-            audioSource = mPacketSources.valueFor(STREAMTYPE_AUDIO);
-            resumeMask |= STREAMTYPE_AUDIO;
-        }
-
-        sp<AnotherPacketSource> videoSource;
-        if ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI) {
-            videoSource = mPacketSources.valueFor(STREAMTYPE_VIDEO);
-            resumeMask |= STREAMTYPE_VIDEO;
-        }
-
-        sp<AnotherPacketSource> subtitleSource;
-        if ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI) {
-            subtitleSource = mPacketSources.valueFor(STREAMTYPE_SUBTITLES);
-            resumeMask |= STREAMTYPE_SUBTITLES;
+        sp<AnotherPacketSource> sources[kMaxStreams];
+        for (size_t j = 0; j < kMaxStreams; ++j) {
+            if ((streamMask & indexToType(j)) && uri == mStreams[j].mUri) {
+                sources[j] = mPacketSources.valueFor(indexToType(j));
+                resumeMask |= indexToType(j);
+            }
         }
 
         CHECK_NE(resumeMask, 0u);
@@ -1053,7 +1020,7 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
         streamMask &= ~resumeMask;
 
         mFetcherInfos.valueAt(i).mFetcher->startAsync(
-                audioSource, videoSource, subtitleSource);
+                sources[kAudioIndex], sources[kVideoIndex], sources[kSubtitleIndex]);
     }
 
     // streamMask now only contains the types that need a new fetcher created.
@@ -1062,52 +1029,33 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
         ALOGV("creating new fetchers for mask 0x%08x", streamMask);
     }
 
-    while (streamMask != 0) {
-        StreamType streamType = (StreamType)(streamMask & ~(streamMask - 1));
+    for (size_t i = 0; i < kMaxStreams; i++) {
+        if (!(indexToType(i) & streamMask)) {
+            continue;
+        }
 
         AString uri;
-        switch (streamType) {
-            case STREAMTYPE_AUDIO:
-                uri = audioURI;
-                break;
-            case STREAMTYPE_VIDEO:
-                uri = videoURI;
-                break;
-            case STREAMTYPE_SUBTITLES:
-                uri = subtitleURI;
-                break;
-            default:
-                TRESPASS();
-        }
+        uri = mStreams[i].mUri;
 
         sp<PlaylistFetcher> fetcher = addFetcher(uri.c_str());
         CHECK(fetcher != NULL);
 
-        sp<AnotherPacketSource> audioSource;
-        if ((streamMask & STREAMTYPE_AUDIO) && uri == audioURI) {
-            audioSource = mPacketSources.valueFor(STREAMTYPE_AUDIO);
-            audioSource->clear();
-
-            streamMask &= ~STREAMTYPE_AUDIO;
-        }
-
-        sp<AnotherPacketSource> videoSource;
-        if ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI) {
-            videoSource = mPacketSources.valueFor(STREAMTYPE_VIDEO);
-            videoSource->clear();
+        sp<AnotherPacketSource> sources[kMaxStreams];
+        // TRICKY: looping from i as earlier streams are already removed from streamMask
+        for (size_t j = i; j < kMaxStreams; ++j) {
+            if ((streamMask & indexToType(j)) && uri == mStreams[j].mUri) {
+                sources[j] = mPacketSources.valueFor(indexToType(j));
+                sources[j]->clear();
 
-            streamMask &= ~STREAMTYPE_VIDEO;
-        }
-
-        sp<AnotherPacketSource> subtitleSource;
-        if ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI) {
-            subtitleSource = mPacketSources.valueFor(STREAMTYPE_SUBTITLES);
-            subtitleSource->clear();
-
-            streamMask &= ~STREAMTYPE_SUBTITLES;
+                streamMask &= ~indexToType(j);
+            }
         }
 
-        fetcher->startAsync(audioSource, videoSource, subtitleSource, timeUs);
+        fetcher->startAsync(
+                sources[kAudioIndex],
+                sources[kVideoIndex],
+                sources[kSubtitleIndex],
+                timeUs);
     }
 
     // All fetchers have now been started, the configuration change
diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h
index f42d7ae..b38c518 100644
--- a/media/libstagefright/httplive/LiveSession.h
+++ b/media/libstagefright/httplive/LiveSession.h
@@ -42,10 +42,17 @@ struct LiveSession : public AHandler {
             const sp<AMessage> &notify,
             uint32_t flags = 0, bool uidValid = false, uid_t uid = 0);
 
+    enum StreamIndex {
+        kAudioIndex    = 0,
+        kVideoIndex    = 1,
+        kSubtitleIndex = 2,
+        kMaxStreams    = 3,
+    };
+
     enum StreamType {
-        STREAMTYPE_AUDIO        = 1,
-        STREAMTYPE_VIDEO        = 2,
-        STREAMTYPE_SUBTITLES    = 4,
+        STREAMTYPE_AUDIO        = 1 << kAudioIndex,
+        STREAMTYPE_VIDEO        = 1 << kVideoIndex,
+        STREAMTYPE_SUBTITLES    = 1 << kSubtitleIndex,
     };
     status_t dequeueAccessUnit(StreamType stream, sp<ABuffer> *accessUnit);
 
@@ -105,6 +112,19 @@ private:
         bool mIsPrepared;
     };
 
+    struct StreamItem {
+        const char *mType;
+        AString mUri;
+        StreamItem() : mType("") {}
+        StreamItem(const char *type) : mType(type) {}
+        AString uriKey() {
+            AString key(mType);
+            key.append("URI");
+            return key;
+        }
+    };
+    StreamItem mStreams[kMaxStreams];
+
     sp<AMessage> mNotify;
     uint32_t mFlags;
     bool mUIDValid;
@@ -123,7 +143,6 @@ private:
     sp<M3UParser> mPlaylist;
 
     KeyedVector<AString, FetcherInfo> mFetcherInfos;
-    AString mAudioURI, mVideoURI, mSubtitleURI;
     uint32_t mStreamMask;
 
     KeyedVector<StreamType, sp<AnotherPacketSource> > mPacketSources;
@@ -171,6 +190,7 @@ private:
     size_t getBandwidthIndex();
 
     static int SortByBandwidth(const BandwidthItem *, const BandwidthItem *);
+    static StreamType indexToType(int idx);
 
     void changeConfiguration(
             int64_t timeUs, size_t bandwidthIndex, bool pickTrack = false);
diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp
index ae19ffa..a9184b5 100644
--- a/media/libstagefright/httplive/M3UParser.cpp
+++ b/media/libstagefright/httplive/M3UParser.cpp
@@ -388,18 +388,6 @@ bool M3UParser::getTypeURI(size_t index, const char *key, AString *uri) const {
     return true;
 }
 
-bool M3UParser::getAudioURI(size_t index, AString *uri) const {
-    return getTypeURI(index, "audio", uri);
-}
-
-bool M3UParser::getVideoURI(size_t index, AString *uri) const {
-    return getTypeURI(index, "video", uri);
-}
-
-bool M3UParser::getSubtitleURI(size_t index, AString *uri) const {
-    return getTypeURI(index, "subtitles", uri);
-}
-
 static bool MakeURL(const char *baseURL, const char *url, AString *out) {
     out->clear();
 
diff --git a/media/libstagefright/httplive/M3UParser.h b/media/libstagefright/httplive/M3UParser.h
index b93b0e5..ccd6556 100644
--- a/media/libstagefright/httplive/M3UParser.h
+++ b/media/libstagefright/httplive/M3UParser.h
@@ -45,9 +45,7 @@ struct M3UParser : public RefBase {
     status_t getTrackInfo(Parcel* reply) const;
     ssize_t getSelectedIndex() const;
 
-    bool getAudioURI(size_t index, AString *uri) const;
-    bool getVideoURI(size_t index, AString *uri) const;
-    bool getSubtitleURI(size_t index, AString *uri) const;
+    bool getTypeURI(size_t index, const char *key, AString *uri) const;
 
 protected:
     virtual ~M3UParser();
@@ -95,8 +93,6 @@ private:
 
     status_t parseMedia(const AString &line);
 
-    bool getTypeURI(size_t index, const char *key, AString *uri) const;
-
     static status_t ParseInt32(const char *s, int32_t *x);
     static status_t ParseDouble(const char *s, double *x);
 
-- 
cgit v1.1


From 777ee5ed736c8f6c3f7d196ea022f7432bfd23e1 Mon Sep 17 00:00:00 2001
From: Robert Shih <robertshih@google.com>
Date: Thu, 20 Feb 2014 13:07:26 -0800
Subject: Initial HLS seamless switch implementation.

Bug: 11854054
Change-Id: I75fc2a258111295039ac13cc37e407df25891dd2
---
 media/libstagefright/httplive/LiveSession.cpp     | 280 +++++++++++++++++++---
 media/libstagefright/httplive/LiveSession.h       |  36 +++
 media/libstagefright/httplive/PlaylistFetcher.cpp | 251 +++++++++++++++++--
 media/libstagefright/httplive/PlaylistFetcher.h   |  22 +-
 4 files changed, 541 insertions(+), 48 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index 79ad100..9e30ebd 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -37,6 +37,8 @@
 #include <media/stagefright/MetaData.h>
 #include <media/stagefright/Utils.h>
 
+#include <utils/Mutex.h>
+
 #include <ctype.h>
 #include <openssl/aes.h>
 #include <openssl/md5.h>
@@ -57,10 +59,14 @@ LiveSession::LiveSession(
                     : 0)),
       mPrevBandwidthIndex(-1),
       mStreamMask(0),
+      mNewStreamMask(0),
+      mSwapMask(0),
       mCheckBandwidthGeneration(0),
+      mSwitchGeneration(0),
       mLastDequeuedTimeUs(0ll),
       mRealTimeBaseUs(0ll),
       mReconfigurationInProgress(false),
+      mSwitchInProgress(false),
       mDisconnectReplyID(0) {
     if (mUIDValid) {
         mHTTPDataSource->setUID(mUID);
@@ -72,16 +78,37 @@ LiveSession::LiveSession(
 
     for (size_t i = 0; i < kMaxStreams; ++i) {
         mPacketSources.add(indexToType(i), new AnotherPacketSource(NULL /* meta */));
+        mPacketSources2.add(indexToType(i), new AnotherPacketSource(NULL /* meta */));
     }
 }
 
 LiveSession::~LiveSession() {
 }
 
+sp<ABuffer> LiveSession::createFormatChangeBuffer(bool swap) {
+    ABuffer *discontinuity = new ABuffer(0);
+    discontinuity->meta()->setInt32("discontinuity", ATSParser::DISCONTINUITY_FORMATCHANGE);
+    discontinuity->meta()->setInt32("swapPacketSource", swap);
+    discontinuity->meta()->setInt32("switchGeneration", mSwitchGeneration);
+    discontinuity->meta()->setInt64("timeUs", -1);
+    return discontinuity;
+}
+
+void LiveSession::swapPacketSource(StreamType stream) {
+    sp<AnotherPacketSource> &aps = mPacketSources.editValueFor(stream);
+    sp<AnotherPacketSource> &aps2 = mPacketSources2.editValueFor(stream);
+    sp<AnotherPacketSource> tmp = aps;
+    aps = aps2;
+    aps2 = tmp;
+    aps2->clear();
+}
+
 status_t LiveSession::dequeueAccessUnit(
         StreamType stream, sp<ABuffer> *accessUnit) {
     if (!(mStreamMask & stream)) {
-        return UNKNOWN_ERROR;
+        // return -EWOULDBLOCK to avoid halting the decoder
+        // when switching between audio/video and audio only.
+        return -EWOULDBLOCK;
     }
 
     sp<AnotherPacketSource> packetSource = mPacketSources.valueFor(stream);
@@ -121,6 +148,25 @@ status_t LiveSession::dequeueAccessUnit(
               streamStr,
               type,
               extra == NULL ? "NULL" : extra->debugString().c_str());
+
+        int32_t swap;
+        if (type == ATSParser::DISCONTINUITY_FORMATCHANGE
+                && (*accessUnit)->meta()->findInt32("swapPacketSource", &swap)
+                && swap) {
+
+            int32_t switchGeneration;
+            CHECK((*accessUnit)->meta()->findInt32("switchGeneration", &switchGeneration));
+            {
+                Mutex::Autolock lock(mSwapMutex);
+                if (switchGeneration == mSwitchGeneration) {
+                    swapPacketSource(stream);
+                    sp<AMessage> msg = new AMessage(kWhatSwapped, id());
+                    msg->setInt32("stream", stream);
+                    msg->setInt32("switchGeneration", switchGeneration);
+                    msg->post();
+                }
+            }
+        }
     } else if (err == OK) {
         if (stream == STREAMTYPE_AUDIO || stream == STREAMTYPE_VIDEO) {
             int64_t timeUs;
@@ -142,6 +188,7 @@ status_t LiveSession::dequeueAccessUnit(
 }
 
 status_t LiveSession::getStreamFormat(StreamType stream, sp<AMessage> *format) {
+    // No swapPacketSource race condition; called from the same thread as dequeueAccessUnit.
     if (!(mStreamMask & stream)) {
         return UNKNOWN_ERROR;
     }
@@ -238,7 +285,12 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) {
                     if (what == PlaylistFetcher::kWhatStopped) {
                         AString uri;
                         CHECK(msg->findString("uri", &uri));
-                        mFetcherInfos.removeItem(uri);
+                        if (mFetcherInfos.removeItem(uri) < 0) {
+                            // ignore duplicated kWhatStopped messages.
+                            break;
+                        }
+
+                        tryToFinishBandwidthSwitch();
                     }
 
                     if (mContinuation != NULL) {
@@ -274,6 +326,8 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) {
                         postPrepared(err);
                     }
 
+                    cancelBandwidthSwitch();
+
                     mPacketSources.valueFor(STREAMTYPE_AUDIO)->signalEOS(err);
 
                     mPacketSources.valueFor(STREAMTYPE_VIDEO)->signalEOS(err);
@@ -312,6 +366,27 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) {
                     break;
                 }
 
+                case PlaylistFetcher::kWhatStartedAt:
+                {
+                    int32_t switchGeneration;
+                    CHECK(msg->findInt32("switchGeneration", &switchGeneration));
+
+                    if (switchGeneration != mSwitchGeneration) {
+                        break;
+                    }
+
+                    // 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.
+                    for (size_t i = 0; i < mFetcherInfos.size(); i++) {
+                        const FetcherInfo info = mFetcherInfos.valueAt(i);
+                        if (info.mToBeRemoved) {
+                            info.mFetcher->resumeUntilAsync(msg);
+                        }
+                    }
+                    break;
+                }
+
                 default:
                     TRESPASS();
             }
@@ -356,6 +431,11 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) {
             break;
         }
 
+        case kWhatSwapped:
+        {
+            onSwapped(msg);
+            break;
+        }
         default:
             TRESPASS();
             break;
@@ -466,6 +546,10 @@ void LiveSession::finishDisconnect() {
     // during disconnection either.
     cancelCheckBandwidthEvent();
 
+    // Protect mPacketSources from a swapPacketSource race condition through disconnect.
+    // (finishDisconnect, onFinishDisconnect2)
+    cancelBandwidthSwitch();
+
     for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
         mFetcherInfos.valueAt(i).mFetcher->stopAsync();
     }
@@ -505,11 +589,13 @@ sp<PlaylistFetcher> LiveSession::addFetcher(const char *uri) {
 
     sp<AMessage> notify = new AMessage(kWhatFetcherNotify, id());
     notify->setString("uri", uri);
+    notify->setInt32("switchGeneration", mSwitchGeneration);
 
     FetcherInfo info;
     info.mFetcher = new PlaylistFetcher(notify, this, uri);
     info.mDurationUs = -1ll;
     info.mIsPrepared = false;
+    info.mToBeRemoved = false;
     looper()->registerHandler(info.mFetcher);
 
     mFetcherInfos.add(uri, info);
@@ -844,8 +930,25 @@ status_t LiveSession::selectTrack(size_t index, bool select) {
     return err;
 }
 
+bool LiveSession::canSwitchUp() {
+    // Allow upwards bandwidth switch when a stream has buffered at least 10 seconds.
+    status_t err = OK;
+    for (size_t i = 0; i < mPacketSources.size(); ++i) {
+        sp<AnotherPacketSource> source = mPacketSources.valueAt(i);
+        int64_t dur = source->getBufferedDurationUs(&err);
+        if (err == OK && dur > 10000000) {
+            return true;
+        }
+    }
+    return false;
+}
+
 void LiveSession::changeConfiguration(
         int64_t timeUs, size_t bandwidthIndex, bool pickTrack) {
+    // Protect mPacketSources from a swapPacketSource race condition through reconfiguration.
+    // (changeConfiguration, onChangeConfiguration2, onChangeConfiguration3).
+    cancelBandwidthSwitch();
+
     CHECK(!mReconfigurationInProgress);
     mReconfigurationInProgress = true;
 
@@ -861,7 +964,8 @@ void LiveSession::changeConfiguration(
     CHECK_LT(bandwidthIndex, mBandwidthItems.size());
     const BandwidthItem &item = mBandwidthItems.itemAt(bandwidthIndex);
 
-    uint32_t streamMask = 0;
+    uint32_t streamMask = 0; // streams that should be fetched by the new fetcher
+    uint32_t resumeMask = 0; // streams that should be fetched by the original fetcher
 
     AString URIs[kMaxStreams];
     for (size_t i = 0; i < kMaxStreams; ++i) {
@@ -879,9 +983,14 @@ void LiveSession::changeConfiguration(
 
         // If we're seeking all current fetchers are discarded.
         if (timeUs < 0ll) {
+            // delay fetcher removal
+            discardFetcher = false;
+
             for (size_t j = 0; j < kMaxStreams; ++j) {
-                if ((streamMask & indexToType(j)) && uri == URIs[j]) {
-                    discardFetcher = false;
+                StreamType type = indexToType(j);
+                if ((streamMask & type) && uri == URIs[j]) {
+                    resumeMask |= type;
+                    streamMask &= ~type;
                 }
             }
         }
@@ -893,8 +1002,15 @@ void LiveSession::changeConfiguration(
         }
     }
 
-    sp<AMessage> msg = new AMessage(kWhatChangeConfiguration2, id());
+    sp<AMessage> msg;
+    if (timeUs < 0ll) {
+        // skip onChangeConfiguration2 (decoder destruction) if switching.
+        msg = new AMessage(kWhatChangeConfiguration3, id());
+    } else {
+        msg = new AMessage(kWhatChangeConfiguration2, id());
+    }
     msg->setInt32("streamMask", streamMask);
+    msg->setInt32("resumeMask", resumeMask);
     msg->setInt64("timeUs", timeUs);
     for (size_t i = 0; i < kMaxStreams; ++i) {
         if (streamMask & indexToType(i)) {
@@ -977,11 +1093,13 @@ void LiveSession::onChangeConfiguration2(const sp<AMessage> &msg) {
 }
 
 void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
+    mContinuation.clear();
     // All remaining fetchers are still suspended, the player has shutdown
     // any decoders that needed it.
 
-    uint32_t streamMask;
+    uint32_t streamMask, resumeMask;
     CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask));
+    CHECK(msg->findInt32("resumeMask", (int32_t *)&resumeMask));
 
     for (size_t i = 0; i < kMaxStreams; ++i) {
         if (streamMask & indexToType(i)) {
@@ -990,37 +1108,39 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
     }
 
     int64_t timeUs;
+    bool switching = false;
     CHECK(msg->findInt64("timeUs", &timeUs));
 
     if (timeUs < 0ll) {
         timeUs = mLastDequeuedTimeUs;
+        switching = true;
     }
     mRealTimeBaseUs = ALooper::GetNowUs() - timeUs;
 
-    mStreamMask = streamMask;
+    mNewStreamMask = streamMask;
 
-    // Resume all existing fetchers and assign them packet sources.
+    // Of all existing fetchers:
+    // * Resume fetchers that are still needed and assign them original packet sources.
+    // * Mark otherwise unneeded fetchers for removal.
+    ALOGV("resuming fetchers for mask 0x%08x", resumeMask);
     for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
         const AString &uri = mFetcherInfos.keyAt(i);
 
-        uint32_t resumeMask = 0;
-
         sp<AnotherPacketSource> sources[kMaxStreams];
         for (size_t j = 0; j < kMaxStreams; ++j) {
-            if ((streamMask & indexToType(j)) && uri == mStreams[j].mUri) {
+            if ((resumeMask & indexToType(j)) && uri == mStreams[j].mUri) {
                 sources[j] = mPacketSources.valueFor(indexToType(j));
-                resumeMask |= indexToType(j);
             }
         }
 
-        CHECK_NE(resumeMask, 0u);
-
-        ALOGV("resuming fetchers for mask 0x%08x", resumeMask);
-
-        streamMask &= ~resumeMask;
-
-        mFetcherInfos.valueAt(i).mFetcher->startAsync(
-                sources[kAudioIndex], sources[kVideoIndex], sources[kSubtitleIndex]);
+        FetcherInfo &info = mFetcherInfos.editValueAt(i);
+        if (sources[kAudioIndex] != NULL || sources[kVideoIndex] != NULL
+                || sources[kSubtitleIndex] != NULL) {
+            info.mFetcher->startAsync(
+                    sources[kAudioIndex], sources[kVideoIndex], sources[kSubtitleIndex]);
+        } else {
+            info.mToBeRemoved = true;
+        }
     }
 
     // streamMask now only contains the types that need a new fetcher created.
@@ -1029,6 +1149,8 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
         ALOGV("creating new fetchers for mask 0x%08x", streamMask);
     }
 
+    // Find out when the original fetchers have buffered up to and start the new fetchers
+    // at a later timestamp.
     for (size_t i = 0; i < kMaxStreams; i++) {
         if (!(indexToType(i) & streamMask)) {
             continue;
@@ -1040,12 +1162,40 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
         sp<PlaylistFetcher> fetcher = addFetcher(uri.c_str());
         CHECK(fetcher != NULL);
 
+        int32_t latestSeq = -1;
+        int64_t latestTimeUs = 0ll;
         sp<AnotherPacketSource> sources[kMaxStreams];
+
         // TRICKY: looping from i as earlier streams are already removed from streamMask
         for (size_t j = i; j < kMaxStreams; ++j) {
             if ((streamMask & indexToType(j)) && uri == mStreams[j].mUri) {
                 sources[j] = mPacketSources.valueFor(indexToType(j));
-                sources[j]->clear();
+
+                if (!switching) {
+                    sources[j]->clear();
+                } else {
+                    int32_t type, seq;
+                    int64_t srcTimeUs;
+                    sp<AMessage> meta = sources[j]->getLatestMeta();
+
+                    if (meta != NULL && !meta->findInt32("discontinuity", &type)) {
+                        CHECK(meta->findInt32("seq", &seq));
+                        if (seq > latestSeq) {
+                            latestSeq = seq;
+                        }
+                        CHECK(meta->findInt64("timeUs", &srcTimeUs));
+                        if (srcTimeUs > latestTimeUs) {
+                            latestTimeUs = srcTimeUs;
+                        }
+                    }
+
+                    sources[j] = mPacketSources2.valueFor(indexToType(j));
+                    sources[j]->clear();
+                    uint32_t extraStreams = mNewStreamMask & (~mStreamMask);
+                    if (extraStreams & indexToType(j)) {
+                        sources[j]->queueAccessUnit(createFormatChangeBuffer(/* swap = */ false));
+                    }
+                }
 
                 streamMask &= ~indexToType(j);
             }
@@ -1055,7 +1205,9 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
                 sources[kAudioIndex],
                 sources[kVideoIndex],
                 sources[kSubtitleIndex],
-                timeUs);
+                timeUs,
+                latestTimeUs /* min start time(us) */,
+                latestSeq >= 0 ? latestSeq + 1 : -1 /* starting sequence number hint */ );
     }
 
     // All fetchers have now been started, the configuration change
@@ -1064,14 +1216,61 @@ void LiveSession::onChangeConfiguration3(const sp<AMessage> &msg) {
     scheduleCheckBandwidthEvent();
 
     ALOGV("XXX configuration change completed.");
-
     mReconfigurationInProgress = false;
+    if (switching) {
+        mSwitchInProgress = true;
+    } else {
+        mStreamMask = mNewStreamMask;
+    }
 
     if (mDisconnectReplyID != 0) {
         finishDisconnect();
     }
 }
 
+void LiveSession::onSwapped(const sp<AMessage> &msg) {
+    int32_t switchGeneration;
+    CHECK(msg->findInt32("switchGeneration", &switchGeneration));
+    if (switchGeneration != mSwitchGeneration) {
+        return;
+    }
+
+    int32_t stream;
+    CHECK(msg->findInt32("stream", &stream));
+    mSwapMask |= stream;
+    if (mSwapMask != mStreamMask) {
+        return;
+    }
+
+    // Check if new variant contains extra streams.
+    uint32_t extraStreams = mNewStreamMask & (~mStreamMask);
+    while (extraStreams) {
+        StreamType extraStream = (StreamType) (extraStreams & ~(extraStreams - 1));
+        swapPacketSource(extraStream);
+        extraStreams &= ~extraStream;
+    }
+
+    tryToFinishBandwidthSwitch();
+}
+
+// Mark switch done when:
+//   1. all old buffers are swapped out, AND
+//   2. all old fetchers are removed.
+void LiveSession::tryToFinishBandwidthSwitch() {
+    bool needToRemoveFetchers = false;
+    for (size_t i = 0; i < mFetcherInfos.size(); ++i) {
+        if (mFetcherInfos.valueAt(i).mToBeRemoved) {
+            needToRemoveFetchers = true;
+            break;
+        }
+    }
+    if (!needToRemoveFetchers && mSwapMask == mStreamMask) {
+        mStreamMask = mNewStreamMask;
+        mSwitchInProgress = false;
+        mSwapMask = 0;
+    }
+}
+
 void LiveSession::scheduleCheckBandwidthEvent() {
     sp<AMessage> msg = new AMessage(kWhatCheckBandwidth, id());
     msg->setInt32("generation", mCheckBandwidthGeneration);
@@ -1082,16 +1281,37 @@ void LiveSession::cancelCheckBandwidthEvent() {
     ++mCheckBandwidthGeneration;
 }
 
-void LiveSession::onCheckBandwidth() {
-    if (mReconfigurationInProgress) {
-        scheduleCheckBandwidthEvent();
-        return;
+void LiveSession::cancelBandwidthSwitch() {
+    Mutex::Autolock lock(mSwapMutex);
+    mSwitchGeneration++;
+    mSwitchInProgress = false;
+    mSwapMask = 0;
+}
+
+bool LiveSession::canSwitchBandwidthTo(size_t bandwidthIndex) {
+    if (mReconfigurationInProgress || mSwitchInProgress) {
+        return false;
+    }
+
+    if (mPrevBandwidthIndex < 0) {
+        return true;
     }
 
+    if (bandwidthIndex == (size_t)mPrevBandwidthIndex) {
+        return false;
+    } else if (bandwidthIndex > (size_t)mPrevBandwidthIndex) {
+        return canSwitchUp();
+    } else {
+        return true;
+    }
+}
+
+void LiveSession::onCheckBandwidth() {
     size_t bandwidthIndex = getBandwidthIndex();
-    if (mPrevBandwidthIndex < 0
-            || bandwidthIndex != (size_t)mPrevBandwidthIndex) {
+    if (canSwitchBandwidthTo(bandwidthIndex)) {
         changeConfiguration(-1ll /* timeUs */, bandwidthIndex);
+    } else {
+        scheduleCheckBandwidthEvent();
     }
 
     // Handling the kWhatCheckBandwidth even here does _not_ automatically
diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h
index b38c518..e26d024 100644
--- a/media/libstagefright/httplive/LiveSession.h
+++ b/media/libstagefright/httplive/LiveSession.h
@@ -81,6 +81,11 @@ struct LiveSession : public AHandler {
         kWhatPreparationFailed,
     };
 
+    // create a format-change discontinuity
+    //
+    // swap:
+    //   whether is format-change discontinuity should trigger a buffer swap
+    sp<ABuffer> createFormatChangeBuffer(bool swap = true);
 protected:
     virtual ~LiveSession();
 
@@ -99,6 +104,7 @@ private:
         kWhatChangeConfiguration2       = 'chC2',
         kWhatChangeConfiguration3       = 'chC3',
         kWhatFinishDisconnect2          = 'fin2',
+        kWhatSwapped                    = 'swap',
     };
 
     struct BandwidthItem {
@@ -110,6 +116,7 @@ private:
         sp<PlaylistFetcher> mFetcher;
         int64_t mDurationUs;
         bool mIsPrepared;
+        bool mToBeRemoved;
     };
 
     struct StreamItem {
@@ -145,9 +152,26 @@ private:
     KeyedVector<AString, FetcherInfo> mFetcherInfos;
     uint32_t mStreamMask;
 
+    // Masks used during reconfiguration:
+    // mNewStreamMask: streams in the variant playlist we're switching to;
+    // we don't want to immediately overwrite the original value.
+    uint32_t mNewStreamMask;
+
+    // mSwapMask: streams that have started to playback content in the new variant playlist;
+    // we use this to track reconfiguration progress.
+    uint32_t mSwapMask;
+
     KeyedVector<StreamType, sp<AnotherPacketSource> > mPacketSources;
+    // A second set of packet sources that buffer content for the variant we're switching to.
+    KeyedVector<StreamType, sp<AnotherPacketSource> > mPacketSources2;
+
+    // A mutex used to serialize two sets of events:
+    // * the swapping of packet sources in dequeueAccessUnit on the player thread, AND
+    // * a forced bandwidth switch termination in cancelSwitch on the live looper.
+    Mutex mSwapMutex;
 
     int32_t mCheckBandwidthGeneration;
+    int32_t mSwitchGeneration;
 
     size_t mContinuationCounter;
     sp<AMessage> mContinuation;
@@ -156,6 +180,7 @@ private:
     int64_t mRealTimeBaseUs;
 
     bool mReconfigurationInProgress;
+    bool mSwitchInProgress;
     uint32_t mDisconnectReplyID;
 
     sp<PlaylistFetcher> addFetcher(const char *uri);
@@ -197,16 +222,27 @@ private:
     void onChangeConfiguration(const sp<AMessage> &msg);
     void onChangeConfiguration2(const sp<AMessage> &msg);
     void onChangeConfiguration3(const sp<AMessage> &msg);
+    void onSwapped(const sp<AMessage> &msg);
+    void tryToFinishBandwidthSwitch();
 
     void scheduleCheckBandwidthEvent();
     void cancelCheckBandwidthEvent();
 
+    // cancelBandwidthSwitch is atomic wrt swapPacketSource; call it to prevent packet sources
+    // from being swapped out on stale discontinuities while manipulating
+    // mPacketSources/mPacketSources2.
+    void cancelBandwidthSwitch();
+
+    bool canSwitchBandwidthTo(size_t bandwidthIndex);
     void onCheckBandwidth();
 
     void finishDisconnect();
 
     void postPrepared(status_t err);
 
+    void swapPacketSource(StreamType stream);
+    bool canSwitchUp();
+
     DISALLOW_EVIL_CONSTRUCTORS(LiveSession);
 };
 
diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index 9cb16fe..0eac8b3 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -48,16 +48,20 @@ namespace android {
 // static
 const int64_t PlaylistFetcher::kMinBufferedDurationUs = 10000000ll;
 const int64_t PlaylistFetcher::kMaxMonitorDelayUs = 3000000ll;
+const int32_t PlaylistFetcher::kNumSkipFrames = 10;
 
 PlaylistFetcher::PlaylistFetcher(
         const sp<AMessage> &notify,
         const sp<LiveSession> &session,
         const char *uri)
     : mNotify(notify),
+      mStartTimeUsNotify(notify->dup()),
       mSession(session),
       mURI(uri),
       mStreamTypeMask(0),
       mStartTimeUs(-1ll),
+      mMinStartTimeUs(0ll),
+      mStopParams(NULL),
       mLastPlaylistFetchTimeUs(-1ll),
       mSeqNumber(-1),
       mNumRetries(0),
@@ -69,6 +73,8 @@ PlaylistFetcher::PlaylistFetcher(
       mFirstPTSValid(false),
       mAbsoluteTimeAnchorUs(0ll) {
     memset(mPlaylistHash, 0, sizeof(mPlaylistHash));
+    mStartTimeUsNotify->setInt32("what", kWhatStartedAt);
+    mStartTimeUsNotify->setInt32("streamMask", 0);
 }
 
 PlaylistFetcher::~PlaylistFetcher() {
@@ -325,7 +331,9 @@ void PlaylistFetcher::startAsync(
         const sp<AnotherPacketSource> &audioSource,
         const sp<AnotherPacketSource> &videoSource,
         const sp<AnotherPacketSource> &subtitleSource,
-        int64_t startTimeUs) {
+        int64_t startTimeUs,
+        int64_t minStartTimeUs,
+        int32_t startSeqNumberHint) {
     sp<AMessage> msg = new AMessage(kWhatStart, id());
 
     uint32_t streamTypeMask = 0ul;
@@ -347,6 +355,8 @@ void PlaylistFetcher::startAsync(
 
     msg->setInt32("streamTypeMask", streamTypeMask);
     msg->setInt64("startTimeUs", startTimeUs);
+    msg->setInt64("minStartTimeUs", minStartTimeUs);
+    msg->setInt32("startSeqNumberHint", startSeqNumberHint);
     msg->post();
 }
 
@@ -358,6 +368,12 @@ void PlaylistFetcher::stopAsync() {
     (new AMessage(kWhatStop, id()))->post();
 }
 
+void PlaylistFetcher::resumeUntilAsync(const sp<AMessage> &params) {
+    AMessage* msg = new AMessage(kWhatResumeUntil, id());
+    msg->setMessage("params", params);
+    msg->post();
+}
+
 void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) {
     switch (msg->what()) {
         case kWhatStart:
@@ -392,6 +408,7 @@ void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) {
         }
 
         case kWhatMonitorQueue:
+        case kWhatDownloadNext:
         {
             int32_t generation;
             CHECK(msg->findInt32("generation", &generation));
@@ -401,7 +418,17 @@ void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) {
                 break;
             }
 
-            onMonitorQueue();
+            if (msg->what() == kWhatMonitorQueue) {
+                onMonitorQueue();
+            } else {
+                onDownloadNext();
+            }
+            break;
+        }
+
+        case kWhatResumeUntil:
+        {
+            onResumeUntil(msg);
             break;
         }
 
@@ -417,7 +444,10 @@ status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) {
     CHECK(msg->findInt32("streamTypeMask", (int32_t *)&streamTypeMask));
 
     int64_t startTimeUs;
+    int32_t startSeqNumberHint;
     CHECK(msg->findInt64("startTimeUs", &startTimeUs));
+    CHECK(msg->findInt64("minStartTimeUs", (int64_t *) &mMinStartTimeUs));
+    CHECK(msg->findInt32("startSeqNumberHint", &startSeqNumberHint));
 
     if (streamTypeMask & LiveSession::STREAMTYPE_AUDIO) {
         void *ptr;
@@ -455,6 +485,10 @@ status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) {
         mPrepared = false;
     }
 
+    if (startSeqNumberHint >= 0) {
+        mSeqNumber = startSeqNumberHint;
+    }
+
     postMonitorQueue();
 
     return OK;
@@ -462,20 +496,70 @@ status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) {
 
 void PlaylistFetcher::onPause() {
     cancelMonitorQueue();
+}
+
+void PlaylistFetcher::onStop() {
+    cancelMonitorQueue();
 
     mPacketSources.clear();
     mStreamTypeMask = 0;
 }
 
-void PlaylistFetcher::onStop() {
-    cancelMonitorQueue();
+// Resume until we have reached the boundary timestamps listed in `msg`; when
+// the remaining time is too short (within a resume threshold) stop immediately
+// instead.
+status_t PlaylistFetcher::onResumeUntil(const sp<AMessage> &msg) {
+    sp<AMessage> params;
+    CHECK(msg->findMessage("params", &params));
+
+    bool stop = false;
+    for (size_t i = 0; i < mPacketSources.size(); i++) {
+        sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
+
+        const char *stopKey;
+        int streamType = mPacketSources.keyAt(i);
+        switch (streamType) {
+        case LiveSession::STREAMTYPE_VIDEO:
+            stopKey = "timeUsVideo";
+            break;
 
-    for (size_t i = 0; i < mPacketSources.size(); ++i) {
-        mPacketSources.valueAt(i)->clear();
+        case LiveSession::STREAMTYPE_AUDIO:
+            stopKey = "timeUsAudio";
+            break;
+
+        case LiveSession::STREAMTYPE_SUBTITLES:
+            stopKey = "timeUsSubtitle";
+            break;
+
+        default:
+            TRESPASS();
+        }
+
+        // Don't resume if we would stop within a resume threshold.
+        int64_t latestTimeUs = 0, stopTimeUs = 0;
+        sp<AMessage> latestMeta = packetSource->getLatestMeta();
+        if (latestMeta != NULL
+                && (latestMeta->findInt64("timeUs", &latestTimeUs)
+                && params->findInt64(stopKey, &stopTimeUs))) {
+            int64_t diffUs = stopTimeUs - latestTimeUs;
+            if (diffUs < resumeThreshold(latestMeta)) {
+                stop = true;
+            }
+        }
     }
 
-    mPacketSources.clear();
-    mStreamTypeMask = 0;
+    if (stop) {
+        for (size_t i = 0; i < mPacketSources.size(); i++) {
+            mPacketSources.valueAt(i)->queueAccessUnit(mSession->createFormatChangeBuffer());
+        }
+        stopAsync();
+        return OK;
+    }
+
+    mStopParams = params;
+    postMonitorQueue();
+
+    return OK;
 }
 
 void PlaylistFetcher::notifyError(status_t err) {
@@ -519,8 +603,9 @@ void PlaylistFetcher::onMonitorQueue() {
                 packetSource->getBufferedDurationUs(&finalResult);
         finalResult = OK;
     } else {
-        bool first = true;
-
+        // 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.
         for (size_t i = 0; i < mPacketSources.size(); ++i) {
             if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0) {
                 continue;
@@ -528,9 +613,10 @@ void PlaylistFetcher::onMonitorQueue() {
 
             int64_t bufferedStreamDurationUs =
                 mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult);
-            if (first || bufferedStreamDurationUs < bufferedDurationUs) {
+            ALOGV("buffered %lld for stream %d",
+                    bufferedStreamDurationUs, mPacketSources.keyAt(i));
+            if (bufferedStreamDurationUs > bufferedDurationUs) {
                 bufferedDurationUs = bufferedStreamDurationUs;
-                first = false;
             }
         }
     }
@@ -550,7 +636,12 @@ void PlaylistFetcher::onMonitorQueue() {
     if (finalResult == OK && downloadMore) {
         ALOGV("monitoring, buffered=%lld < %lld",
                 bufferedDurationUs, durationToBufferUs);
-        onDownloadNext();
+        // delay the next download slightly; hopefully this gives other concurrent fetchers
+        // a better chance to run.
+        // onDownloadNext();
+        sp<AMessage> msg = new AMessage(kWhatDownloadNext, id());
+        msg->setInt32("generation", mMonitorQueueGeneration);
+        msg->post(1000l);
     } else {
         // Nothing to do yet, try again in a second.
 
@@ -617,6 +708,12 @@ void PlaylistFetcher::onDownloadNext() {
     const int32_t lastSeqNumberInPlaylist =
         firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1;
 
+    if (mStartup && mSeqNumber >= 0
+            && (mSeqNumber < firstSeqNumberInPlaylist || mSeqNumber > lastSeqNumberInPlaylist)) {
+        // in case we guessed wrong during reconfiguration, try fetching the latest content.
+        mSeqNumber = lastSeqNumberInPlaylist;
+    }
+
     if (mSeqNumber < 0) {
         CHECK_GE(mStartTimeUs, 0ll);
 
@@ -761,6 +858,18 @@ void PlaylistFetcher::onDownloadNext() {
 
     err = extractAndQueueAccessUnits(buffer, itemMeta);
 
+    if (err == -EAGAIN) {
+        // bad starting sequence number hint
+        postMonitorQueue();
+        return;
+    }
+
+    if (err == ERROR_OUT_OF_RANGE) {
+        // reached stopping point
+        stopAsync();
+        return;
+    }
+
     if (err != OK) {
         notifyError(err);
         return;
@@ -817,12 +926,15 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits(
         }
 
         if (mTSParser == NULL) {
-            mTSParser = new ATSParser;
+            // Use TS_TIMESTAMPS_ARE_ABSOLUTE so pts carry over between fetchers.
+            mTSParser = new ATSParser(ATSParser::TS_TIMESTAMPS_ARE_ABSOLUTE);
         }
 
         if (mNextPTSTimeUs >= 0ll) {
             sp<AMessage> extra = new AMessage;
-            extra->setInt64(IStreamListener::kKeyMediaTimeUs, mNextPTSTimeUs);
+            // Since we are using absolute timestamps, signal an offset of 0 to prevent
+            // ATSParser from skewing the timestamps of access units.
+            extra->setInt64(IStreamListener::kKeyMediaTimeUs, 0);
 
             mTSParser->signalDiscontinuity(
                     ATSParser::DISCONTINUITY_SEEK, extra);
@@ -841,17 +953,23 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits(
             offset += 188;
         }
 
+        status_t err = OK;
         for (size_t i = mPacketSources.size(); i-- > 0;) {
             sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
 
+            const char *key;
             ATSParser::SourceType type;
-            switch (mPacketSources.keyAt(i)) {
+            const LiveSession::StreamType stream = mPacketSources.keyAt(i);
+            switch (stream) {
+
                 case LiveSession::STREAMTYPE_VIDEO:
                     type = ATSParser::VIDEO;
+                    key = "timeUsVideo";
                     break;
 
                 case LiveSession::STREAMTYPE_AUDIO:
                     type = ATSParser::AUDIO;
+                    key = "timeUsAudio";
                     break;
 
                 case LiveSession::STREAMTYPE_SUBTITLES:
@@ -878,19 +996,87 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits(
                 continue;
             }
 
+            int64_t timeUs;
             sp<ABuffer> accessUnit;
             status_t finalResult;
             while (source->hasBufferAvailable(&finalResult)
                     && source->dequeueAccessUnit(&accessUnit) == OK) {
-                // Note that we do NOT dequeue any discontinuities.
+
+                CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
+                if (mMinStartTimeUs > 0) {
+                    if (timeUs < mMinStartTimeUs) {
+                        // TODO untested path
+                        // try a later ts
+                        int32_t targetDuration;
+                        mPlaylist->meta()->findInt32("target-duration", &targetDuration);
+                        int32_t incr = (mMinStartTimeUs - timeUs) / 1000000 / targetDuration;
+                        if (incr == 0) {
+                            // increment mSeqNumber by at least one
+                            incr = 1;
+                        }
+                        mSeqNumber += incr;
+                        err = -EAGAIN;
+                        break;
+                    } else {
+                        int64_t startTimeUs;
+                        if (mStartTimeUsNotify != NULL
+                                && !mStartTimeUsNotify->findInt64(key, &startTimeUs)) {
+                            mStartTimeUsNotify->setInt64(key, timeUs);
+
+                            uint32_t streamMask = 0;
+                            mStartTimeUsNotify->findInt32("streamMask", (int32_t *) &streamMask);
+                            streamMask |= mPacketSources.keyAt(i);
+                            mStartTimeUsNotify->setInt32("streamMask", streamMask);
+
+                            if (streamMask == mStreamTypeMask) {
+                                mStartTimeUsNotify->post();
+                                mStartTimeUsNotify.clear();
+                            }
+                        }
+                    }
+                }
+
+                if (mStopParams != NULL) {
+                    // Queue discontinuity in original stream.
+                    int64_t stopTimeUs;
+                    if (!mStopParams->findInt64(key, &stopTimeUs) || timeUs >= stopTimeUs) {
+                        packetSource->queueAccessUnit(mSession->createFormatChangeBuffer());
+                        mStreamTypeMask &= ~stream;
+                        mPacketSources.removeItemsAt(i);
+                        break;
+                    }
+                }
+
+                // Note that we do NOT dequeue any discontinuities except for format change.
 
                 // for simplicity, store a reference to the format in each unit
                 sp<MetaData> format = source->getFormat();
                 if (format != NULL) {
                     accessUnit->meta()->setObject("format", format);
                 }
+
+                // Stash the sequence number so we can hint future fetchers where to start at.
+                accessUnit->meta()->setInt32("seq", mSeqNumber);
                 packetSource->queueAccessUnit(accessUnit);
             }
+
+            if (err != OK) {
+                break;
+            }
+        }
+
+        if (err != OK) {
+            for (size_t i = mPacketSources.size(); i-- > 0;) {
+                sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
+                packetSource->clear();
+            }
+            return err;
+        }
+
+        if (!mStreamTypeMask) {
+            // Signal gap is filled between original and new stream.
+            ALOGV("ERROR OUT OF RANGE");
+            return ERROR_OUT_OF_RANGE;
         }
 
         return OK;
@@ -907,6 +1093,7 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits(
         CHECK(itemMeta->findInt64("durationUs", &durationUs));
         buffer->meta()->setInt64("timeUs", getSegmentStartTimeUs(mSeqNumber));
         buffer->meta()->setInt64("durationUs", durationUs);
+        buffer->meta()->setInt32("seq", mSeqNumber);
 
         packetSource->queueAccessUnit(buffer);
         return OK;
@@ -1043,6 +1230,7 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits(
         // Each AAC frame encodes 1024 samples.
         numSamples += 1024;
 
+        unit->meta()->setInt32("seq", mSeqNumber);
         packetSource->queueAccessUnit(unit);
 
         offset += aac_frame_length;
@@ -1070,4 +1258,33 @@ void PlaylistFetcher::updateDuration() {
     msg->post();
 }
 
+int64_t PlaylistFetcher::resumeThreshold(const sp<AMessage> &msg) {
+    int64_t durationUs, threshold;
+    if (msg->findInt64("durationUs", &durationUs)) {
+        return kNumSkipFrames * durationUs;
+    }
+
+    sp<RefBase> obj;
+    msg->findObject("format", &obj);
+    MetaData *format = static_cast<MetaData *>(obj.get());
+
+    const char *mime;
+    CHECK(format->findCString(kKeyMIMEType, &mime));
+    bool audio = !strncasecmp(mime, "audio/", 6);
+    if (audio) {
+        // Assumes 1000 samples per frame.
+        int32_t sampleRate;
+        CHECK(format->findInt32(kKeySampleRate, &sampleRate));
+        return kNumSkipFrames  /* frames */ * 1000 /* samples */
+                * (1000000 / sampleRate) /* sample duration (us) */;
+    } else {
+        int32_t frameRate;
+        if (format->findInt32(kKeyFrameRate, &frameRate) && frameRate > 0) {
+            return kNumSkipFrames * (1000000 / frameRate);
+        }
+    }
+
+    return 500000ll;
+}
+
 }  // namespace android
diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h
index ac04a77..2e0349f 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.h
+++ b/media/libstagefright/httplive/PlaylistFetcher.h
@@ -43,6 +43,7 @@ struct PlaylistFetcher : public AHandler {
         kWhatTemporarilyDoneFetching,
         kWhatPrepared,
         kWhatPreparationFailed,
+        kWhatStartedAt,
     };
 
     PlaylistFetcher(
@@ -56,12 +57,16 @@ struct PlaylistFetcher : public AHandler {
             const sp<AnotherPacketSource> &audioSource,
             const sp<AnotherPacketSource> &videoSource,
             const sp<AnotherPacketSource> &subtitleSource,
-            int64_t startTimeUs = -1ll);
+            int64_t startTimeUs = -1ll,
+            int64_t minStartTimeUs = 0ll /* start after this timestamp */,
+            int32_t startSeqNumberHint = -1 /* try starting at this sequence number */);
 
     void pauseAsync();
 
     void stopAsync();
 
+    void resumeUntilAsync(const sp<AMessage> &params);
+
 protected:
     virtual ~PlaylistFetcher();
     virtual void onMessageReceived(const sp<AMessage> &msg);
@@ -76,17 +81,25 @@ private:
         kWhatPause          = 'paus',
         kWhatStop           = 'stop',
         kWhatMonitorQueue   = 'moni',
+        kWhatResumeUntil    = 'rsme',
+        kWhatDownloadNext   = 'dlnx',
     };
 
     static const int64_t kMinBufferedDurationUs;
     static const int64_t kMaxMonitorDelayUs;
+    static const int32_t kNumSkipFrames;
 
+    // notifications to mSession
     sp<AMessage> mNotify;
+    sp<AMessage> mStartTimeUsNotify;
+
     sp<LiveSession> mSession;
     AString mURI;
 
     uint32_t mStreamTypeMask;
     int64_t mStartTimeUs;
+    int64_t mMinStartTimeUs; // start fetching no earlier than this value
+    sp<AMessage> mStopParams; // message containing the latest timestamps we should fetch.
 
     KeyedVector<LiveSession::StreamType, sp<AnotherPacketSource> >
         mPacketSources;
@@ -153,6 +166,9 @@ private:
     void onMonitorQueue();
     void onDownloadNext();
 
+    // Resume a fetcher to continue until the stopping point stored in msg.
+    status_t onResumeUntil(const sp<AMessage> &msg);
+
     status_t extractAndQueueAccessUnits(
             const sp<ABuffer> &buffer, const sp<AMessage> &itemMeta);
 
@@ -165,6 +181,10 @@ private:
 
     void updateDuration();
 
+    // Before resuming a fetcher in onResume, check the remaining duration is longer than that
+    // returned by resumeThreshold.
+    int64_t resumeThreshold(const sp<AMessage> &msg);
+
     DISALLOW_EVIL_CONSTRUCTORS(PlaylistFetcher);
 };
 
-- 
cgit v1.1


From e2492dc46f5476e3abc617ee21d74f44198591dd Mon Sep 17 00:00:00 2001
From: Martin Storsjo <martin@martin.st>
Date: Fri, 22 Nov 2013 18:11:58 +0200
Subject: ChromiumHTTPDataSource: Keep track of the redirected URL

This makes the code actually match an existing comment in
DrmInitialization, which claimed that mURI was the redirected
URL and not the original one.

Bug: 13174301
Change-Id: I0a5cc65f520f1482ff91320ae78af84a8a681ee3
---
 media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp | 5 +++++
 media/libstagefright/chromium_http/support.cpp                | 1 +
 media/libstagefright/include/ChromiumHTTPDataSource.h         | 1 +
 3 files changed, 7 insertions(+)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp b/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp
index a862d8b..7e5c280 100644
--- a/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp
+++ b/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp
@@ -108,6 +108,11 @@ status_t ChromiumHTTPDataSource::connect_l(
     return mState == CONNECTED ? OK : mIOResult;
 }
 
+void ChromiumHTTPDataSource::onRedirect(const char *url) {
+    Mutex::Autolock autoLock(mLock);
+    mURI = url;
+}
+
 void ChromiumHTTPDataSource::onConnectionEstablished(
         int64_t contentSize, const char *contentType) {
     Mutex::Autolock autoLock(mLock);
diff --git a/media/libstagefright/chromium_http/support.cpp b/media/libstagefright/chromium_http/support.cpp
index 0a8e3e3..3b33212 100644
--- a/media/libstagefright/chromium_http/support.cpp
+++ b/media/libstagefright/chromium_http/support.cpp
@@ -269,6 +269,7 @@ bool SfDelegate::getUID(uid_t *uid) const {
 void SfDelegate::OnReceivedRedirect(
             net::URLRequest *request, const GURL &new_url, bool *defer_redirect) {
     MY_LOGV("OnReceivedRedirect");
+    mOwner->onRedirect(new_url.spec().c_str());
 }
 
 void SfDelegate::OnAuthRequired(
diff --git a/media/libstagefright/include/ChromiumHTTPDataSource.h b/media/libstagefright/include/ChromiumHTTPDataSource.h
index 785f939..da188dd 100644
--- a/media/libstagefright/include/ChromiumHTTPDataSource.h
+++ b/media/libstagefright/include/ChromiumHTTPDataSource.h
@@ -113,6 +113,7 @@ private:
     void onConnectionFailed(status_t err);
     void onReadCompleted(ssize_t size);
     void onDisconnectComplete();
+    void onRedirect(const char *url);
 
     void clearDRMState_l();
 
-- 
cgit v1.1


From be1195a6d5e6dd4299da344cf9905dd6a12dc4ef Mon Sep 17 00:00:00 2001
From: Marco Nelissen <marcone@google.com>
Date: Mon, 25 Nov 2013 08:39:56 -0800
Subject: Fix SIGABRT when playing mp4 file

If the track duration was shorter than the segment duration, the calculated
encoder padding would be negative, resulting in a crash.
b/11823061
https://code.google.com/p/android/issues/detail?id=62610

Change-Id: I3989ad88caea38d212b61355c15aec13382c6116
---
 media/libstagefright/MPEG4Extractor.cpp | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 6f51c39..9f95212 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -960,6 +960,12 @@ status_t MPEG4Extractor::parseChunk(off64_t *offset, int depth) {
                     mLastTrack->meta->setInt32(kKeyEncoderDelay, delay);
 
                     int64_t paddingus = duration - (segment_duration + media_time);
+                    if (paddingus < 0) {
+                        // track duration from media header (which is what kKeyDuration is) might
+                        // be slightly shorter than the segment duration, which would make the
+                        // padding negative. Clamp to zero.
+                        paddingus = 0;
+                    }
                     int64_t paddingsamples = (paddingus * samplerate + 500000) / 1000000;
                     mLastTrack->meta->setInt32(kKeyEncoderPadding, paddingsamples);
                 }
-- 
cgit v1.1


From 75d03185ac7be95c700f3f375080989e5de03ef6 Mon Sep 17 00:00:00 2001
From: Martin Storsjo <martin@martin.st>
Date: Fri, 22 Nov 2013 17:05:05 +0200
Subject: M3UParser: Skip query strings when looking for the last slash in a
 URL

Bug: 13174301
Change-Id: I72d3a5e11fef9bbd75b291bc490c9cab1dce58da
---
 media/libstagefright/httplive/M3UParser.cpp | 34 +++++++++++++++++++----------
 1 file changed, 22 insertions(+), 12 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp
index a9184b5..8f530ee 100644
--- a/media/libstagefright/httplive/M3UParser.cpp
+++ b/media/libstagefright/httplive/M3UParser.cpp
@@ -423,22 +423,32 @@ static bool MakeURL(const char *baseURL, const char *url, AString *out) {
     } else {
         // URL is a relative path
 
-        size_t n = strlen(baseURL);
-        if (baseURL[n - 1] == '/') {
-            out->setTo(baseURL);
-            out->append(url);
+        // Check for a possible query string
+        const char *qsPos = strchr(baseURL, '?');
+        size_t end;
+        if (qsPos != NULL) {
+            end = qsPos - baseURL;
         } else {
-            const char *slashPos = strrchr(baseURL, '/');
-
-            if (slashPos > &baseURL[6]) {
-                out->setTo(baseURL, slashPos - baseURL);
-            } else {
-                out->setTo(baseURL);
+            end = strlen(baseURL);
+        }
+        // Check for the last slash before a potential query string
+        for (ssize_t pos = end - 1; pos >= 0; pos--) {
+            if (baseURL[pos] == '/') {
+                end = pos;
+                break;
             }
+        }
 
-            out->append("/");
-            out->append(url);
+        // Check whether the found slash actually is part of the path
+        // and not part of the "http://".
+        if (end > 6) {
+            out->setTo(baseURL, end);
+        } else {
+            out->setTo(baseURL);
         }
+
+        out->append("/");
+        out->append(url);
     }
 
     ALOGV("base:'%s', url:'%s' => '%s'", baseURL, url, out->c_str());
-- 
cgit v1.1


From a1076eb135b74a32e9bdc1aed17aee4374eb41af Mon Sep 17 00:00:00 2001
From: Martin Storsjo <martin@martin.st>
Date: Fri, 22 Nov 2013 17:57:31 +0200
Subject: LiveSession: Use the actual, possibly redirected url as base in the
 M3U

This fixes playback of HLS streams where the M3U playlists use
relative paths and the main playlist is opened via a redirect.

Bug: 13174301
Change-Id: I787e1c050daddc566be4c8e3f84803d3f91eec46
---
 media/libstagefright/httplive/LiveSession.cpp | 14 +++++++++++---
 media/libstagefright/httplive/LiveSession.h   |  3 ++-
 2 files changed, 13 insertions(+), 4 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index 9e30ebd..687e871 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -623,7 +623,8 @@ status_t LiveSession::fetchFile(
         const char *url, sp<ABuffer> *out,
         int64_t range_offset, int64_t range_length,
         uint32_t block_size, /* download block size */
-        sp<DataSource> *source /* to return and reuse source */) {
+        sp<DataSource> *source, /* to return and reuse source */
+        String8 *actualUrl) {
     off64_t size;
     sp<DataSource> temp_source;
     if (source == NULL) {
@@ -719,6 +720,12 @@ status_t LiveSession::fetchFile(
     }
 
     *out = buffer;
+    if (actualUrl != NULL) {
+        *actualUrl = (*source)->getUri();
+        if (actualUrl->isEmpty()) {
+            *actualUrl = url;
+        }
+    }
 
     return OK;
 }
@@ -730,7 +737,8 @@ sp<M3UParser> LiveSession::fetchPlaylist(
     *unchanged = false;
 
     sp<ABuffer> buffer;
-    status_t err = fetchFile(url, &buffer);
+    String8 actualUrl;
+    status_t err = fetchFile(url, &buffer, 0, -1, 0, NULL, &actualUrl);
 
     if (err != OK) {
         return NULL;
@@ -764,7 +772,7 @@ sp<M3UParser> LiveSession::fetchPlaylist(
 #endif
 
     sp<M3UParser> playlist =
-        new M3UParser(url, buffer->data(), buffer->size());
+        new M3UParser(actualUrl.string(), buffer->data(), buffer->size());
 
     if (playlist->initCheck() != OK) {
         ALOGE("failed to parse .m3u8 playlist");
diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h
index e26d024..376d451 100644
--- a/media/libstagefright/httplive/LiveSession.h
+++ b/media/libstagefright/httplive/LiveSession.h
@@ -207,7 +207,8 @@ private:
             /* download block size */
             uint32_t block_size = 0,
             /* reuse DataSource if doing partial fetch */
-            sp<DataSource> *source = NULL);
+            sp<DataSource> *source = NULL,
+            String8 *actualUrl = NULL);
 
     sp<M3UParser> fetchPlaylist(
             const char *url, uint8_t *curPlaylistHash, bool *unchanged);
-- 
cgit v1.1


From 5874615f26a6ba9cda8ba359e5035b21f6e3818b Mon Sep 17 00:00:00 2001
From: Marco Nelissen <marcone@google.com>
Date: Mon, 3 Feb 2014 14:33:16 -0800
Subject: Allow for larger codec private data

This change adds support for multibyte sizes in the ESDS. Previously the
Matroska extractor only supported single byte sizes, so codec private data
had to be less than 108 bytes, and would crash if it was bigger.
b/12584090

Change-Id: I9edfc2f687187d1e98bcfd2fe56576312435df3e
---
 .../libstagefright/matroska/MatroskaExtractor.cpp  | 80 ++++++++++++++--------
 1 file changed, 50 insertions(+), 30 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/matroska/MatroskaExtractor.cpp b/media/libstagefright/matroska/MatroskaExtractor.cpp
index d260d0f..dcb1cda 100644
--- a/media/libstagefright/matroska/MatroskaExtractor.cpp
+++ b/media/libstagefright/matroska/MatroskaExtractor.cpp
@@ -716,41 +716,61 @@ bool MatroskaExtractor::isLiveStreaming() const {
     return mIsLiveStreaming;
 }
 
+static int bytesForSize(size_t size) {
+    // use at most 28 bits (4 times 7)
+    CHECK(size <= 0xfffffff);
+
+    if (size > 0x1fffff) {
+        return 4;
+    } else if (size > 0x3fff) {
+        return 3;
+    } else if (size > 0x7f) {
+        return 2;
+    }
+    return 1;
+}
+
+static void storeSize(uint8_t *data, size_t &idx, size_t size) {
+    int numBytes = bytesForSize(size);
+    idx += numBytes;
+
+    data += idx;
+    size_t next = 0;
+    while (numBytes--) {
+        *--data = (size & 0x7f) | next;
+        size >>= 7;
+        next = 0x80;
+    }
+}
+
 static void addESDSFromCodecPrivate(
         const sp<MetaData> &meta,
         bool isAudio, const void *priv, size_t privSize) {
-    static const uint8_t kStaticESDS[] = {
-        0x03, 22,
-        0x00, 0x00,     // ES_ID
-        0x00,           // streamDependenceFlag, URL_Flag, OCRstreamFlag
-
-        0x04, 17,
-        0x40,           // ObjectTypeIndication
-        0x00, 0x00, 0x00, 0x00,
-        0x00, 0x00, 0x00, 0x00,
-        0x00, 0x00, 0x00, 0x00,
-
-        0x05,
-        // CodecSpecificInfo (with size prefix) follows
-    };
 
-    // Make sure all sizes can be coded in a single byte.
-    CHECK(privSize + 22 - 2 < 128);
-    size_t esdsSize = sizeof(kStaticESDS) + privSize + 1;
+    int privSizeBytesRequired = bytesForSize(privSize);
+    int esdsSize2 = 14 + privSizeBytesRequired + privSize;
+    int esdsSize2BytesRequired = bytesForSize(esdsSize2);
+    int esdsSize1 = 4 + esdsSize2BytesRequired + esdsSize2;
+    int esdsSize1BytesRequired = bytesForSize(esdsSize1);
+    size_t esdsSize = 1 + esdsSize1BytesRequired + esdsSize1;
     uint8_t *esds = new uint8_t[esdsSize];
-    memcpy(esds, kStaticESDS, sizeof(kStaticESDS));
-    uint8_t *ptr = esds + sizeof(kStaticESDS);
-    *ptr++ = privSize;
-    memcpy(ptr, priv, privSize);
-
-    // Increment by codecPrivateSize less 2 bytes that are accounted for
-    // already in lengths of 22/17
-    esds[1] += privSize - 2;
-    esds[6] += privSize - 2;
-
-    // Set ObjectTypeIndication.
-    esds[7] = isAudio ? 0x40   // Audio ISO/IEC 14496-3
-                      : 0x20;  // Visual ISO/IEC 14496-2
+
+    size_t idx = 0;
+    esds[idx++] = 0x03;
+    storeSize(esds, idx, esdsSize1);
+    esds[idx++] = 0x00; // ES_ID
+    esds[idx++] = 0x00; // ES_ID
+    esds[idx++] = 0x00; // streamDependenceFlag, URL_Flag, OCRstreamFlag
+    esds[idx++] = 0x04;
+    storeSize(esds, idx, esdsSize2);
+    esds[idx++] = isAudio ? 0x40   // Audio ISO/IEC 14496-3
+                          : 0x20;  // Visual ISO/IEC 14496-2
+    for (int i = 0; i < 12; i++) {
+        esds[idx++] = 0x00;
+    }
+    esds[idx++] = 0x05;
+    storeSize(esds, idx, privSize);
+    memcpy(esds + idx, priv, privSize);
 
     meta->setData(kKeyESDS, 0, esds, esdsSize);
 
-- 
cgit v1.1


From 1c608ab33acab82d4a87f1abedcc99a623d1b53a Mon Sep 17 00:00:00 2001
From: Chong Zhang <chz@google.com>
Date: Fri, 17 Jan 2014 16:29:59 -0800
Subject: save mGraphicBuffer pointer even when we're suspended

Bug: 12609966
Change-Id: Ifd41fd973876da69039113fcaeacdccf39472b37
(cherry picked from commit 49270665e7a20cd120724fc388da8b166ff0b4f1)
---
 media/libstagefright/omx/GraphicBufferSource.cpp | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/omx/GraphicBufferSource.cpp b/media/libstagefright/omx/GraphicBufferSource.cpp
index b8970ad..7492577 100644
--- a/media/libstagefright/omx/GraphicBufferSource.cpp
+++ b/media/libstagefright/omx/GraphicBufferSource.cpp
@@ -609,6 +609,12 @@ void GraphicBufferSource::onFrameAvailable() {
         BufferQueue::BufferItem item;
         status_t err = mBufferQueue->acquireBuffer(&item, 0);
         if (err == OK) {
+            // If this is the first time we're seeing this buffer, add it to our
+            // slot table.
+            if (item.mGraphicBuffer != NULL) {
+                ALOGV("fillCodecBuffer_l: setting mBufferSlot %d", item.mBuf);
+                mBufferSlot[item.mBuf] = item.mGraphicBuffer;
+            }
             mBufferQueue->releaseBuffer(item.mBuf, item.mFrameNumber,
                     EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, item.mFence);
         }
-- 
cgit v1.1


From 8bd4d16aa5636e98522c07ae31236420788aa749 Mon Sep 17 00:00:00 2001
From: Chong Zhang <chz@google.com>
Date: Fri, 10 Jan 2014 17:36:57 -0800
Subject: Cap pts gap between adjacent frames to specified value

- In the scenario of cast mirroring, encoding could be suspended
  for prolonged periods. Limiting the pts gap to workaround the
  problem where encoder's rate control logic produces huge frames
  after a long period of suspension.

- Repeat last frame a couple more times to get better quality
  on static scenes.

- Fix the timestamp on repeat frames (it was not set)

Bug: 11971963
Change-Id: I1d68ab3d269874bf3921aa429a985c5f63e428c7
(cherry picked from commit 94ee4b708acfa941581160b267afb79192b1d816)
---
 media/libstagefright/ACodec.cpp                  |  22 ++++-
 media/libstagefright/omx/GraphicBufferSource.cpp | 100 ++++++++++++++++++++++-
 media/libstagefright/omx/GraphicBufferSource.h   |  25 +++++-
 media/libstagefright/omx/OMXNodeInstance.cpp     |  26 +++++-
 4 files changed, 167 insertions(+), 6 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 3810ac1..9276818 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -371,7 +371,8 @@ ACodec::ACodec()
       mDequeueCounter(0),
       mStoreMetaDataInOutputBuffers(false),
       mMetaDataBuffersToSubmit(0),
-      mRepeatFrameDelayUs(-1ll) {
+      mRepeatFrameDelayUs(-1ll),
+      mMaxPtsGapUs(-1l) {
     mUninitializedState = new UninitializedState(this);
     mLoadedState = new LoadedState(this);
     mLoadedToIdleState = new LoadedToIdleState(this);
@@ -1109,6 +1110,10 @@ status_t ACodec::configureCodec(
                     &mRepeatFrameDelayUs)) {
             mRepeatFrameDelayUs = -1ll;
         }
+
+        if (!msg->findInt64("max-pts-gap-to-encoder", &mMaxPtsGapUs)) {
+            mMaxPtsGapUs = -1l;
+        }
     }
 
     // Always try to enable dynamic output buffers on native surface
@@ -3855,6 +3860,21 @@ void ACodec::LoadedState::onCreateInputSurface(
         }
     }
 
+    if (err == OK && mCodec->mMaxPtsGapUs > 0l) {
+        err = mCodec->mOMX->setInternalOption(
+                mCodec->mNode,
+                kPortIndexInput,
+                IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP,
+                &mCodec->mMaxPtsGapUs,
+                sizeof(mCodec->mMaxPtsGapUs));
+
+        if (err != OK) {
+            ALOGE("[%s] Unable to configure max timestamp gap (err %d)",
+                  mCodec->mComponentName.c_str(),
+                  err);
+        }
+    }
+
     if (err == OK) {
         notify->setObject("input-surface",
                 new BufferProducerWrapper(bufferProducer));
diff --git a/media/libstagefright/omx/GraphicBufferSource.cpp b/media/libstagefright/omx/GraphicBufferSource.cpp
index b8970ad..f1bb4f1 100644
--- a/media/libstagefright/omx/GraphicBufferSource.cpp
+++ b/media/libstagefright/omx/GraphicBufferSource.cpp
@@ -42,7 +42,11 @@ GraphicBufferSource::GraphicBufferSource(OMXNodeInstance* nodeInstance,
     mEndOfStream(false),
     mEndOfStreamSent(false),
     mRepeatAfterUs(-1ll),
+    mMaxTimestampGapUs(-1ll),
+    mPrevOriginalTimeUs(-1ll),
+    mPrevModifiedTimeUs(-1ll),
     mRepeatLastFrameGeneration(0),
+    mRepeatLastFrameTimestamp(-1ll),
     mLatestSubmittedBufferId(-1),
     mLatestSubmittedBufferFrameNum(0),
     mLatestSubmittedBufferUseCount(0),
@@ -299,6 +303,32 @@ void GraphicBufferSource::codecBufferEmptied(OMX_BUFFERHEADERTYPE* header) {
     return;
 }
 
+void GraphicBufferSource::codecBufferFilled(OMX_BUFFERHEADERTYPE* header) {
+    Mutex::Autolock autoLock(mMutex);
+
+    if (mMaxTimestampGapUs > 0ll
+            && !(header->nFlags & OMX_BUFFERFLAG_CODECCONFIG)) {
+        ssize_t index = mOriginalTimeUs.indexOfKey(header->nTimeStamp);
+        if (index >= 0) {
+            ALOGV("OUT timestamp: %lld -> %lld",
+                    header->nTimeStamp, mOriginalTimeUs[index]);
+            header->nTimeStamp = mOriginalTimeUs[index];
+            mOriginalTimeUs.removeItemsAt(index);
+        } else {
+            // giving up the effort as encoder doesn't appear to preserve pts
+            ALOGW("giving up limiting timestamp gap (pts = %lld)",
+                    header->nTimeStamp);
+            mMaxTimestampGapUs = -1ll;
+        }
+        if (mOriginalTimeUs.size() > BufferQueue::NUM_BUFFER_SLOTS) {
+            // something terribly wrong must have happened, giving up...
+            ALOGE("mOriginalTimeUs has too many entries (%d)",
+                    mOriginalTimeUs.size());
+            mMaxTimestampGapUs = -1ll;
+        }
+    }
+}
+
 void GraphicBufferSource::suspend(bool suspend) {
     Mutex::Autolock autoLock(mMutex);
 
@@ -431,6 +461,7 @@ bool GraphicBufferSource::repeatLatestSubmittedBuffer_l() {
     BufferQueue::BufferItem item;
     item.mBuf = mLatestSubmittedBufferId;
     item.mFrameNumber = mLatestSubmittedBufferFrameNum;
+    item.mTimestamp = mRepeatLastFrameTimestamp;
 
     status_t err = submitBuffer_l(item, cbi);
 
@@ -440,6 +471,20 @@ bool GraphicBufferSource::repeatLatestSubmittedBuffer_l() {
 
     ++mLatestSubmittedBufferUseCount;
 
+    /* repeat last frame up to kRepeatLastFrameCount times.
+     * in case of static scene, a single repeat might not get rid of encoder
+     * ghosting completely, refresh a couple more times to get better quality
+     */
+    if (--mRepeatLastFrameCount > 0) {
+        mRepeatLastFrameTimestamp = item.mTimestamp + mRepeatAfterUs * 1000;
+
+        if (mReflector != NULL) {
+            sp<AMessage> msg = new AMessage(kWhatRepeatLastFrame, mReflector->id());
+            msg->setInt32("generation", ++mRepeatLastFrameGeneration);
+            msg->post(mRepeatAfterUs);
+        }
+    }
+
     return true;
 }
 
@@ -460,8 +505,11 @@ void GraphicBufferSource::setLatestSubmittedBuffer_l(
 
     mLatestSubmittedBufferId = item.mBuf;
     mLatestSubmittedBufferFrameNum = item.mFrameNumber;
+    mRepeatLastFrameTimestamp = item.mTimestamp + mRepeatAfterUs * 1000;
+
     mLatestSubmittedBufferUseCount = 1;
     mRepeatBufferDeferred = false;
+    mRepeatLastFrameCount = kRepeatLastFrameCount;
 
     if (mReflector != NULL) {
         sp<AMessage> msg = new AMessage(kWhatRepeatLastFrame, mReflector->id());
@@ -497,6 +545,39 @@ status_t GraphicBufferSource::signalEndOfInputStream() {
     return OK;
 }
 
+int64_t GraphicBufferSource::getTimestamp(const BufferQueue::BufferItem &item) {
+    int64_t timeUs = item.mTimestamp / 1000;
+
+    if (mMaxTimestampGapUs > 0ll) {
+        /* Cap timestamp gap between adjacent frames to specified max
+         *
+         * In the scenario of cast mirroring, encoding could be suspended for
+         * prolonged periods. Limiting the pts gap to workaround the problem
+         * where encoder's rate control logic produces huge frames after a
+         * long period of suspension.
+         */
+
+        int64_t originalTimeUs = timeUs;
+        if (mPrevOriginalTimeUs >= 0ll) {
+            if (originalTimeUs < mPrevOriginalTimeUs) {
+                // Drop the frame if it's going backward in time. Bad timestamp
+                // could disrupt encoder's rate control completely.
+                ALOGV("Dropping frame that's going backward in time");
+                return -1;
+            }
+            int64_t timestampGapUs = originalTimeUs - mPrevOriginalTimeUs;
+            timeUs = (timestampGapUs < mMaxTimestampGapUs ?
+                    timestampGapUs : mMaxTimestampGapUs) + mPrevModifiedTimeUs;
+        }
+        mPrevOriginalTimeUs = originalTimeUs;
+        mPrevModifiedTimeUs = timeUs;
+        mOriginalTimeUs.add(timeUs, originalTimeUs);
+        ALOGV("IN  timestamp: %lld -> %lld", originalTimeUs, timeUs);
+    }
+
+    return timeUs;
+}
+
 status_t GraphicBufferSource::submitBuffer_l(
         const BufferQueue::BufferItem &item, int cbi) {
     ALOGV("submitBuffer_l cbi=%d", cbi);
@@ -513,9 +594,15 @@ status_t GraphicBufferSource::submitBuffer_l(
     memcpy(data, &type, 4);
     memcpy(data + 4, &handle, sizeof(buffer_handle_t));
 
+    int64_t timeUs = getTimestamp(item);
+    if (timeUs < 0ll) {
+        ALOGE("Dropping frame with bad timestamp");
+        return UNKNOWN_ERROR;
+    }
+
     status_t err = mNodeInstance->emptyDirectBuffer(header, 0,
             4 + sizeof(buffer_handle_t), OMX_BUFFERFLAG_ENDOFFRAME,
-            item.mTimestamp / 1000);
+            timeUs);
     if (err != OK) {
         ALOGW("WARNING: emptyDirectBuffer failed: 0x%x", err);
         codecBuffer.mGraphicBuffer = NULL;
@@ -658,6 +745,17 @@ status_t GraphicBufferSource::setRepeatPreviousFrameDelayUs(
     return OK;
 }
 
+status_t GraphicBufferSource::setMaxTimestampGapUs(int64_t maxGapUs) {
+    Mutex::Autolock autoLock(mMutex);
+
+    if (mExecuting || maxGapUs <= 0ll) {
+        return INVALID_OPERATION;
+    }
+
+    mMaxTimestampGapUs = maxGapUs;
+
+    return OK;
+}
 void GraphicBufferSource::onMessageReceived(const sp<AMessage> &msg) {
     switch (msg->what()) {
         case kWhatRepeatLastFrame:
diff --git a/media/libstagefright/omx/GraphicBufferSource.h b/media/libstagefright/omx/GraphicBufferSource.h
index 9e5eee6..3b0e454 100644
--- a/media/libstagefright/omx/GraphicBufferSource.h
+++ b/media/libstagefright/omx/GraphicBufferSource.h
@@ -87,6 +87,10 @@ public:
     // fill it with a new frame of data; otherwise, just mark it as available.
     void codecBufferEmptied(OMX_BUFFERHEADERTYPE* header);
 
+    // Called when omx_message::FILL_BUFFER_DONE is received. (Currently the
+    // buffer source will fix timestamp in the header if needed.)
+    void codecBufferFilled(OMX_BUFFERHEADERTYPE* header);
+
     // This is called after the last input frame has been submitted.  We
     // need to submit an empty buffer with the EOS flag set.  If we don't
     // have a codec buffer ready, we just set the mEndOfStream flag.
@@ -105,6 +109,15 @@ public:
     // state and once this behaviour is specified it cannot be reset.
     status_t setRepeatPreviousFrameDelayUs(int64_t repeatAfterUs);
 
+    // When set, the timestamp fed to the encoder will be modified such that
+    // the gap between two adjacent frames is capped at maxGapUs. Timestamp
+    // will be restored to the original when the encoded frame is returned to
+    // the client.
+    // This is to solve a problem in certain real-time streaming case, where
+    // encoder's rate control logic produces huge frames after a long period
+    // of suspension on input.
+    status_t setMaxTimestampGapUs(int64_t maxGapUs);
+
 protected:
     // BufferQueue::ConsumerListener interface, called when a new frame of
     // data is available.  If we're executing and a codec buffer is
@@ -165,6 +178,7 @@ private:
 
     void setLatestSubmittedBuffer_l(const BufferQueue::BufferItem &item);
     bool repeatLatestSubmittedBuffer_l();
+    int64_t getTimestamp(const BufferQueue::BufferItem &item);
 
     // Lock, covers all member variables.
     mutable Mutex mMutex;
@@ -206,13 +220,22 @@ private:
     enum {
         kWhatRepeatLastFrame,
     };
-
+    enum {
+        kRepeatLastFrameCount = 10,
+    };
     int64_t mRepeatAfterUs;
+    int64_t mMaxTimestampGapUs;
+
+    KeyedVector<int64_t, int64_t> mOriginalTimeUs;
+    int64_t mPrevOriginalTimeUs;
+    int64_t mPrevModifiedTimeUs;
 
     sp<ALooper> mLooper;
     sp<AHandlerReflector<GraphicBufferSource> > mReflector;
 
     int32_t mRepeatLastFrameGeneration;
+    int64_t mRepeatLastFrameTimestamp;
+    int32_t mRepeatLastFrameCount;
 
     int mLatestSubmittedBufferId;
     uint64_t mLatestSubmittedBufferFrameNum;
diff --git a/media/libstagefright/omx/OMXNodeInstance.cpp b/media/libstagefright/omx/OMXNodeInstance.cpp
index 5f104fc..6c5c857 100644
--- a/media/libstagefright/omx/OMXNodeInstance.cpp
+++ b/media/libstagefright/omx/OMXNodeInstance.cpp
@@ -849,6 +849,7 @@ status_t OMXNodeInstance::setInternalOption(
     switch (type) {
         case IOMX::INTERNAL_OPTION_SUSPEND:
         case IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY:
+        case IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP:
         {
             const sp<GraphicBufferSource> &bufferSource =
                 getGraphicBufferSource();
@@ -864,7 +865,8 @@ status_t OMXNodeInstance::setInternalOption(
 
                 bool suspend = *(bool *)data;
                 bufferSource->suspend(suspend);
-            } else {
+            } else if (type ==
+                    IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY){
                 if (size != sizeof(int64_t)) {
                     return INVALID_OPERATION;
                 }
@@ -872,6 +874,14 @@ status_t OMXNodeInstance::setInternalOption(
                 int64_t delayUs = *(int64_t *)data;
 
                 return bufferSource->setRepeatPreviousFrameDelayUs(delayUs);
+            } else {
+                if (size != sizeof(int64_t)) {
+                    return INVALID_OPERATION;
+                }
+
+                int64_t maxGapUs = *(int64_t *)data;
+
+                return bufferSource->setMaxTimestampGapUs(maxGapUs);
             }
 
             return OK;
@@ -883,6 +893,8 @@ status_t OMXNodeInstance::setInternalOption(
 }
 
 void OMXNodeInstance::onMessage(const omx_message &msg) {
+    const sp<GraphicBufferSource>& bufferSource(getGraphicBufferSource());
+
     if (msg.type == omx_message::FILL_BUFFER_DONE) {
         OMX_BUFFERHEADERTYPE *buffer =
             static_cast<OMX_BUFFERHEADERTYPE *>(
@@ -892,10 +904,18 @@ void OMXNodeInstance::onMessage(const omx_message &msg) {
             static_cast<BufferMeta *>(buffer->pAppPrivate);
 
         buffer_meta->CopyFromOMX(buffer);
-    } else if (msg.type == omx_message::EMPTY_BUFFER_DONE) {
-        const sp<GraphicBufferSource>& bufferSource(getGraphicBufferSource());
 
         if (bufferSource != NULL) {
+            // fix up the buffer info (especially timestamp) if needed
+            bufferSource->codecBufferFilled(buffer);
+
+            omx_message newMsg = msg;
+            newMsg.u.extended_buffer_data.timestamp = buffer->nTimeStamp;
+            mObserver->onMessage(newMsg);
+            return;
+        }
+    } else if (msg.type == omx_message::EMPTY_BUFFER_DONE) {
+        if (bufferSource != NULL) {
             // This is one of the buffers used exclusively by
             // GraphicBufferSource.
             // Don't dispatch a message back to ACodec, since it doesn't
-- 
cgit v1.1


From ceeabe15f4e7bc73efdfcafed917202de9d515cb Mon Sep 17 00:00:00 2001
From: Robert Shih <robertshih@google.com>
Date: Mon, 3 Mar 2014 15:08:27 -0800
Subject: httplive: clear access units before returning from seekTo.

Bug: 12060952
Change-Id: I6a69a718c082501003ee9b78a948a2f8bbfbb14e
---
 media/libstagefright/httplive/LiveSession.cpp     | 26 +++++++++++++++++------
 media/libstagefright/httplive/LiveSession.h       |  2 ++
 media/libstagefright/httplive/PlaylistFetcher.cpp | 25 ++++++++++++++++------
 media/libstagefright/httplive/PlaylistFetcher.h   |  4 ++--
 4 files changed, 42 insertions(+), 15 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index 687e871..c1c3e2b 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -67,7 +67,8 @@ LiveSession::LiveSession(
       mRealTimeBaseUs(0ll),
       mReconfigurationInProgress(false),
       mSwitchInProgress(false),
-      mDisconnectReplyID(0) {
+      mDisconnectReplyID(0),
+      mSeekReplyID(0) {
     if (mUIDValid) {
         mHTTPDataSource->setUID(mUID);
     }
@@ -234,6 +235,10 @@ status_t LiveSession::seekTo(int64_t timeUs) {
     sp<AMessage> response;
     status_t err = msg->postAndAwaitResponse(&response);
 
+    uint32_t replyID;
+    CHECK(response == mSeekReply && 0 != mSeekReplyID);
+    mSeekReply.clear();
+    mSeekReplyID = 0;
     return err;
 }
 
@@ -259,15 +264,12 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) {
 
         case kWhatSeek:
         {
-            uint32_t replyID;
-            CHECK(msg->senderAwaitsResponse(&replyID));
+            CHECK(msg->senderAwaitsResponse(&mSeekReplyID));
 
             status_t err = onSeek(msg);
 
-            sp<AMessage> response = new AMessage;
-            response->setInt32("err", err);
-
-            response->postReply(replyID);
+            mSeekReply = new AMessage;
+            mSeekReply->setInt32("err", err);
             break;
         }
 
@@ -297,6 +299,11 @@ void LiveSession::onMessageReceived(const sp<AMessage> &msg) {
                         CHECK_GT(mContinuationCounter, 0);
                         if (--mContinuationCounter == 0) {
                             mContinuation->post();
+
+                            if (mSeekReplyID != 0) {
+                                CHECK(mSeekReply != NULL);
+                                mSeekReply->postReply(mSeekReplyID);
+                            }
                         }
                     }
                     break;
@@ -1035,6 +1042,11 @@ void LiveSession::changeConfiguration(
 
     if (mContinuationCounter == 0) {
         msg->post();
+
+        if (mSeekReplyID != 0) {
+            CHECK(mSeekReply != NULL);
+            mSeekReply->postReply(mSeekReplyID);
+        }
     }
 }
 
diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h
index 376d451..680792d 100644
--- a/media/libstagefright/httplive/LiveSession.h
+++ b/media/libstagefright/httplive/LiveSession.h
@@ -175,6 +175,7 @@ private:
 
     size_t mContinuationCounter;
     sp<AMessage> mContinuation;
+    sp<AMessage> mSeekReply;
 
     int64_t mLastDequeuedTimeUs;
     int64_t mRealTimeBaseUs;
@@ -182,6 +183,7 @@ private:
     bool mReconfigurationInProgress;
     bool mSwitchInProgress;
     uint32_t mDisconnectReplyID;
+    uint32_t mSeekReplyID;
 
     sp<PlaylistFetcher> addFetcher(const char *uri);
 
diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index 0eac8b3..012a68b 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -364,8 +364,10 @@ void PlaylistFetcher::pauseAsync() {
     (new AMessage(kWhatPause, id()))->post();
 }
 
-void PlaylistFetcher::stopAsync() {
-    (new AMessage(kWhatStop, id()))->post();
+void PlaylistFetcher::stopAsync(bool selfTriggered) {
+    sp<AMessage> msg = new AMessage(kWhatStop, id());
+    msg->setInt32("selfTriggered", selfTriggered);
+    msg->post();
 }
 
 void PlaylistFetcher::resumeUntilAsync(const sp<AMessage> &params) {
@@ -399,7 +401,7 @@ void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) {
 
         case kWhatStop:
         {
-            onStop();
+            onStop(msg);
 
             sp<AMessage> notify = mNotify->dup();
             notify->setInt32("what", kWhatStopped);
@@ -498,9 +500,20 @@ void PlaylistFetcher::onPause() {
     cancelMonitorQueue();
 }
 
-void PlaylistFetcher::onStop() {
+void PlaylistFetcher::onStop(const sp<AMessage> &msg) {
     cancelMonitorQueue();
 
+    int32_t selfTriggered;
+    CHECK(msg->findInt32("selfTriggered", &selfTriggered));
+    if (!selfTriggered) {
+        // Self triggered stops only happen during switching, in which case we do not want
+        // to clear the discontinuities queued at the end of packet sources.
+        for (size_t i = 0; i < mPacketSources.size(); i++) {
+            sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
+            packetSource->clear();
+        }
+    }
+
     mPacketSources.clear();
     mStreamTypeMask = 0;
 }
@@ -552,7 +565,7 @@ status_t PlaylistFetcher::onResumeUntil(const sp<AMessage> &msg) {
         for (size_t i = 0; i < mPacketSources.size(); i++) {
             mPacketSources.valueAt(i)->queueAccessUnit(mSession->createFormatChangeBuffer());
         }
-        stopAsync();
+        stopAsync(/* selfTriggered = */ true);
         return OK;
     }
 
@@ -866,7 +879,7 @@ void PlaylistFetcher::onDownloadNext() {
 
     if (err == ERROR_OUT_OF_RANGE) {
         // reached stopping point
-        stopAsync();
+        stopAsync(/* selfTriggered = */ true);
         return;
     }
 
diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h
index 2e0349f..8404b8d 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.h
+++ b/media/libstagefright/httplive/PlaylistFetcher.h
@@ -63,7 +63,7 @@ struct PlaylistFetcher : public AHandler {
 
     void pauseAsync();
 
-    void stopAsync();
+    void stopAsync(bool selfTriggered = false);
 
     void resumeUntilAsync(const sp<AMessage> &params);
 
@@ -162,7 +162,7 @@ private:
 
     status_t onStart(const sp<AMessage> &msg);
     void onPause();
-    void onStop();
+    void onStop(const sp<AMessage> &msg);
     void onMonitorQueue();
     void onDownloadNext();
 
-- 
cgit v1.1


From 9f434cfd021f60e26baf589dc34bf3839b832d4d Mon Sep 17 00:00:00 2001
From: Robert Shih <robertshih@google.com>
Date: Mon, 10 Mar 2014 17:05:01 -0700
Subject: M3UParser: trim spaces when parsing comma separated codecs.

Bug: 13402087
Change-Id: Idc92716bfefd6d1b0cb371d0d97d990d53288090
---
 media/libstagefright/httplive/M3UParser.cpp | 1 +
 1 file changed, 1 insertion(+)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp
index 8f530ee..f211bc8 100644
--- a/media/libstagefright/httplive/M3UParser.cpp
+++ b/media/libstagefright/httplive/M3UParser.cpp
@@ -365,6 +365,7 @@ bool M3UParser::getTypeURI(size_t index, const char *key, AString *uri) const {
             codecs.append(',');
             while ((commaPos = codecs.find(",", offset)) >= 0) {
                 AString codec(codecs, offset, commaPos - offset);
+                codec.trim();
                 // return true only if a codec of type `key` ("audio"/"video")
                 // is found.
                 if (codecIsType(codec, key)) {
-- 
cgit v1.1


From 7d3044d64294cca6fadd184648a57185e92cf5c6 Mon Sep 17 00:00:00 2001
From: Robert Shih <robertshih@google.com>
Date: Mon, 10 Mar 2014 17:27:15 -0700
Subject: LiveSession: fix incorrect stream key (subtitle"s")

Bug: 13402087
Change-Id: Ic46e3069c6e41f90ead47cae84cbe0123d11002a
---
 media/libstagefright/httplive/LiveSession.cpp | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index c1c3e2b..3e11759 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -75,7 +75,7 @@ LiveSession::LiveSession(
 
     mStreams[kAudioIndex] = StreamItem("audio");
     mStreams[kVideoIndex] = StreamItem("video");
-    mStreams[kSubtitleIndex] = StreamItem("subtitle");
+    mStreams[kSubtitleIndex] = StreamItem("subtitles");
 
     for (size_t i = 0; i < kMaxStreams; ++i) {
         mPacketSources.add(indexToType(i), new AnotherPacketSource(NULL /* meta */));
-- 
cgit v1.1


From cf49a51ff59c3cd228d178d23252ac0d39d5a893 Mon Sep 17 00:00:00 2001
From: Chong Zhang <chz@google.com>
Date: Fri, 14 Feb 2014 15:26:00 -0800
Subject: fix codec buffer leak in error handling when timestamp goes backward

don't touch codec buffer if we decided to drop a frame with bad pts

Bug: 11971963
Change-Id: I9b4c56210f64258f1be257b14184381a1133e0d6
(cherry picked from commit b63d2433350d56bda9f3477549086c90bb6d535e)
---
 media/libstagefright/omx/GraphicBufferSource.cpp | 14 +++++++-------
 1 file changed, 7 insertions(+), 7 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/omx/GraphicBufferSource.cpp b/media/libstagefright/omx/GraphicBufferSource.cpp
index 20fa7ce..44f0be7 100644
--- a/media/libstagefright/omx/GraphicBufferSource.cpp
+++ b/media/libstagefright/omx/GraphicBufferSource.cpp
@@ -562,7 +562,7 @@ int64_t GraphicBufferSource::getTimestamp(const BufferQueue::BufferItem &item) {
             if (originalTimeUs < mPrevOriginalTimeUs) {
                 // Drop the frame if it's going backward in time. Bad timestamp
                 // could disrupt encoder's rate control completely.
-                ALOGV("Dropping frame that's going backward in time");
+                ALOGW("Dropping frame that's going backward in time");
                 return -1;
             }
             int64_t timestampGapUs = originalTimeUs - mPrevOriginalTimeUs;
@@ -581,6 +581,12 @@ int64_t GraphicBufferSource::getTimestamp(const BufferQueue::BufferItem &item) {
 status_t GraphicBufferSource::submitBuffer_l(
         const BufferQueue::BufferItem &item, int cbi) {
     ALOGV("submitBuffer_l cbi=%d", cbi);
+
+    int64_t timeUs = getTimestamp(item);
+    if (timeUs < 0ll) {
+        return UNKNOWN_ERROR;
+    }
+
     CodecBuffer& codecBuffer(mCodecBuffers.editItemAt(cbi));
     codecBuffer.mGraphicBuffer = mBufferSlot[item.mBuf];
     codecBuffer.mBuf = item.mBuf;
@@ -594,12 +600,6 @@ status_t GraphicBufferSource::submitBuffer_l(
     memcpy(data, &type, 4);
     memcpy(data + 4, &handle, sizeof(buffer_handle_t));
 
-    int64_t timeUs = getTimestamp(item);
-    if (timeUs < 0ll) {
-        ALOGE("Dropping frame with bad timestamp");
-        return UNKNOWN_ERROR;
-    }
-
     status_t err = mNodeInstance->emptyDirectBuffer(header, 0,
             4 + sizeof(buffer_handle_t), OMX_BUFFERFLAG_ENDOFFRAME,
             timeUs);
-- 
cgit v1.1


From 8a57d24c733b08da846b54d1adf029e606b5a5f3 Mon Sep 17 00:00:00 2001
From: Marco Nelissen <marcone@google.com>
Date: Thu, 13 Mar 2014 14:18:25 -0700
Subject: Verify certificates

b/13418320

Change-Id: I3052dd5f9ec057e700784cd713f6a7dab9ecfe7b
---
 media/libstagefright/chromium_http/Android.mk  |   1 +
 media/libstagefright/chromium_http/support.cpp | 101 ++++++++++++++++++++++++-
 2 files changed, 101 insertions(+), 1 deletion(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/chromium_http/Android.mk b/media/libstagefright/chromium_http/Android.mk
index f26f386..290427e 100644
--- a/media/libstagefright/chromium_http/Android.mk
+++ b/media/libstagefright/chromium_http/Android.mk
@@ -21,6 +21,7 @@ LOCAL_SHARED_LIBRARIES += \
         libstlport \
         libchromium_net \
         libutils \
+        libbinder \
         libcutils \
         liblog \
         libstagefright_foundation \
diff --git a/media/libstagefright/chromium_http/support.cpp b/media/libstagefright/chromium_http/support.cpp
index 3b33212..d4e82ee 100644
--- a/media/libstagefright/chromium_http/support.cpp
+++ b/media/libstagefright/chromium_http/support.cpp
@@ -40,8 +40,90 @@
 #include <media/stagefright/Utils.h>
 #include <string>
 
+#include <utils/Errors.h>
+#include <binder/IInterface.h>
+#include <binder/IServiceManager.h>
+
 namespace android {
 
+// must be kept in sync with interface defined in IAudioService.aidl
+class IAudioService : public IInterface
+{
+public:
+    DECLARE_META_INTERFACE(AudioService);
+
+    virtual int verifyX509CertChain(
+            const std::vector<std::string>& cert_chain,
+            const std::string& hostname,
+            const std::string& auth_type) = 0;
+};
+
+class BpAudioService : public BpInterface<IAudioService>
+{
+public:
+    BpAudioService(const sp<IBinder>& impl)
+        : BpInterface<IAudioService>(impl)
+    {
+    }
+
+    virtual int verifyX509CertChain(
+            const std::vector<std::string>& cert_chain,
+            const std::string& hostname,
+            const std::string& auth_type)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioService::getInterfaceDescriptor());
+
+        // The vector of std::string we get isn't really a vector of strings,
+        // but rather a vector of binary certificate data. If we try to pass
+        // it to Java language code as a string, it ends up mangled on the other
+        // side, so send them as bytes instead.
+        // Since we can't send an array of byte arrays, send a single array,
+        // which will be split out by the recipient.
+
+        int numcerts = cert_chain.size();
+        data.writeInt32(numcerts);
+        size_t total = 0;
+        for (int i = 0; i < numcerts; i++) {
+            total += cert_chain[i].size();
+        }
+        size_t bytesize = total + numcerts * 4;
+        uint8_t *bytes = (uint8_t*) malloc(bytesize);
+        if (!bytes) {
+            return 5; // SSL_INVALID
+        }
+        ALOGV("%d certs: %d -> %d", numcerts, total, bytesize);
+
+        int offset = 0;
+        for (int i = 0; i < numcerts; i++) {
+            int32_t certsize = cert_chain[i].size();
+            // store this in a known order, which just happens to match the default
+            // byte order of a java ByteBuffer
+            int32_t bigsize = htonl(certsize);
+            ALOGV("cert %d, size %d", i, certsize);
+            memcpy(bytes + offset, &bigsize, sizeof(bigsize));
+            offset += sizeof(bigsize);
+            memcpy(bytes + offset, cert_chain[i].data(), certsize);
+            offset += certsize;
+        }
+        data.writeByteArray(bytesize, bytes);
+        free(bytes);
+        data.writeString16(String16(hostname.c_str()));
+        data.writeString16(String16(auth_type.c_str()));
+
+        int32_t result;
+        if (remote()->transact(IBinder::FIRST_CALL_TRANSACTION, data, &reply) != NO_ERROR
+                || reply.readExceptionCode() < 0 || reply.readInt32(&result) != NO_ERROR) {
+            return 5; // SSL_INVALID;
+        }
+        return result;
+    }
+
+};
+
+IMPLEMENT_META_INTERFACE(AudioService, "android.media.IAudioService");
+
+
 static Mutex gNetworkThreadLock;
 static base::Thread *gNetworkThread = NULL;
 static scoped_refptr<SfRequestContext> gReqContext;
@@ -226,7 +308,24 @@ SfNetworkLibrary::VerifyResult SfNetworkLibrary::VerifyX509CertChain(
         const std::vector<std::string>& cert_chain,
         const std::string& hostname,
         const std::string& auth_type) {
-    return VERIFY_OK;
+
+    sp<IBinder> binder =
+        defaultServiceManager()->checkService(String16("audio"));
+    if (binder == 0) {
+        ALOGW("Thread cannot connect to the audio service");
+    } else {
+        sp<IAudioService> service = interface_cast<IAudioService>(binder);
+        int code = service->verifyX509CertChain(cert_chain, hostname, auth_type);
+        ALOGV("verified: %d", code);
+        if (code == -1) {
+            return VERIFY_OK;
+        } else if (code == 2) { // SSL_IDMISMATCH
+            return VERIFY_BAD_HOSTNAME;
+        } else if (code == 3) { // SSL_UNTRUSTED
+            return VERIFY_NO_TRUSTED_ROOT;
+        }
+    }
+    return VERIFY_INVOCATION_ERROR;
 }
 
 ////////////////////////////////////////////////////////////////////////////////
-- 
cgit v1.1


From 43ca783effd99bba0e6e2dd6fe177a8888578ef8 Mon Sep 17 00:00:00 2001
From: Robert Shih <robertshih@google.com>
Date: Thu, 27 Feb 2014 12:33:24 -0800
Subject: httplive: block-by-block fetch, decrypt, and parse ts files.

Bug: 12060952
Change-Id: I695345081fe23961b9d0ef6db264885f914703ec
---
 media/libstagefright/httplive/LiveSession.cpp     |  10 +-
 media/libstagefright/httplive/LiveSession.h       |   2 +-
 media/libstagefright/httplive/PlaylistFetcher.cpp | 451 +++++++++++++---------
 media/libstagefright/httplive/PlaylistFetcher.h   |   5 +
 media/libstagefright/mpeg2ts/ATSParser.h          |   5 +-
 5 files changed, 285 insertions(+), 188 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index 3e11759..e53e35a 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -626,7 +626,7 @@ sp<PlaylistFetcher> LiveSession::addFetcher(const char *uri) {
  * - block_size == 0 means entire range
  *
  */
-status_t LiveSession::fetchFile(
+ssize_t LiveSession::fetchFile(
         const char *url, sp<ABuffer> *out,
         int64_t range_offset, int64_t range_length,
         uint32_t block_size, /* download block size */
@@ -677,6 +677,7 @@ status_t LiveSession::fetchFile(
         buffer->setRange(0, 0);
     }
 
+    ssize_t bytesRead = 0;
     // adjust range_length if only reading partial block
     if (block_size > 0 && (range_length == -1 || buffer->size() + block_size < range_length)) {
         range_length = buffer->size() + block_size;
@@ -724,6 +725,7 @@ status_t LiveSession::fetchFile(
         }
 
         buffer->setRange(0, buffer->size() + (size_t)n);
+        bytesRead += n;
     }
 
     *out = buffer;
@@ -734,7 +736,7 @@ status_t LiveSession::fetchFile(
         }
     }
 
-    return OK;
+    return bytesRead;
 }
 
 sp<M3UParser> LiveSession::fetchPlaylist(
@@ -745,9 +747,9 @@ sp<M3UParser> LiveSession::fetchPlaylist(
 
     sp<ABuffer> buffer;
     String8 actualUrl;
-    status_t err = fetchFile(url, &buffer, 0, -1, 0, NULL, &actualUrl);
+    ssize_t  err = fetchFile(url, &buffer, 0, -1, 0, NULL, &actualUrl);
 
-    if (err != OK) {
+    if (err <= 0) {
         return NULL;
     }
 
diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h
index 680792d..3f8fee5 100644
--- a/media/libstagefright/httplive/LiveSession.h
+++ b/media/libstagefright/httplive/LiveSession.h
@@ -202,7 +202,7 @@ private:
     //
     // For reused HTTP sources, the caller must download a file sequentially without
     // any overlaps or gaps to prevent reconnection.
-    status_t fetchFile(
+    ssize_t fetchFile(
             const char *url, sp<ABuffer> *out,
             /* request/open a file starting at range_offset for range_length bytes */
             int64_t range_offset = 0, int64_t range_length = -1,
diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index 012a68b..ada856d 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -48,6 +48,7 @@ namespace android {
 // static
 const int64_t PlaylistFetcher::kMinBufferedDurationUs = 10000000ll;
 const int64_t PlaylistFetcher::kMaxMonitorDelayUs = 3000000ll;
+const int32_t PlaylistFetcher::kDownloadBlockSize = 192;
 const int32_t PlaylistFetcher::kNumSkipFrames = 10;
 
 PlaylistFetcher::PlaylistFetcher(
@@ -216,9 +217,9 @@ status_t PlaylistFetcher::decryptBuffer(
     if (index >= 0) {
         key = mAESKeyForURI.valueAt(index);
     } else {
-        status_t err = mSession->fetchFile(keyURI.c_str(), &key);
+        ssize_t err = mSession->fetchFile(keyURI.c_str(), &key);
 
-        if (err != OK) {
+        if (err < 0) {
             ALOGE("failed to fetch cipher key from '%s'.", keyURI.c_str());
             return ERROR_IO;
         } else if (key->size() != 16) {
@@ -704,6 +705,11 @@ status_t PlaylistFetcher::refreshPlaylist() {
     return OK;
 }
 
+// static
+bool PlaylistFetcher::bufferStartsWithTsSyncByte(const sp<ABuffer>& buffer) {
+    return buffer->size() > 0 && buffer->data()[0] == 0x47;
+}
+
 void PlaylistFetcher::onDownloadNext() {
     if (refreshPlaylist() != OK) {
         return;
@@ -823,66 +829,161 @@ void PlaylistFetcher::onDownloadNext() {
 
     ALOGV("fetching '%s'", uri.c_str());
 
-    sp<ABuffer> buffer;
-    status_t err = mSession->fetchFile(
-            uri.c_str(), &buffer, range_offset, range_length);
-
-    if (err != OK) {
-        ALOGE("failed to fetch .ts segment at url '%s'", uri.c_str());
-        notifyError(err);
-        return;
+    sp<DataSource> source;
+    sp<ABuffer> buffer, tsBuffer;
+    // decrypt a junk buffer to prefetch key; since a session uses only one http connection,
+    // this avoids interleaved connections to the key and segment file.
+    {
+        sp<ABuffer> junk = new ABuffer(16);
+        junk->setRange(0, 16);
+        status_t err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, junk,
+                true /* first */);
+        if (err != OK) {
+            notifyError(err);
+            return;
+        }
     }
 
-    CHECK(buffer != NULL);
+    // block-wise download
+    ssize_t bytesRead;
+    do {
+        bytesRead = mSession->fetchFile(
+                uri.c_str(), &buffer, range_offset, range_length, kDownloadBlockSize, &source);
 
-    err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, buffer);
-    if (err == OK) {
-        err = checkDecryptPadding(buffer);
-    }
+        if (bytesRead < 0) {
+            status_t err = bytesRead;
+            ALOGE("failed to fetch .ts segment at url '%s'", uri.c_str());
+            notifyError(err);
+            return;
+        }
 
-    if (err != OK) {
-        ALOGE("decryptBuffer failed w/ error %d", err);
+        CHECK(buffer != NULL);
 
-        notifyError(err);
-        return;
-    }
+        size_t size = buffer->size();
+        // Set decryption range.
+        buffer->setRange(size - bytesRead, bytesRead);
+        status_t err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, buffer,
+                buffer->offset() == 0 /* first */);
+        // Unset decryption range.
+        buffer->setRange(0, size);
 
-    if (mStartup || seekDiscontinuity || explicitDiscontinuity) {
-        // Signal discontinuity.
+        if (err != OK) {
+            ALOGE("decryptBuffer failed w/ error %d", err);
 
-        if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
-            // If this was a live event this made no sense since
-            // we don't have access to all the segment before the current
-            // one.
-            mNextPTSTimeUs = getSegmentStartTimeUs(mSeqNumber);
+            notifyError(err);
+            return;
         }
 
-        if (seekDiscontinuity || explicitDiscontinuity) {
-            ALOGI("queueing discontinuity (seek=%d, explicit=%d)",
-                 seekDiscontinuity, explicitDiscontinuity);
+        if (mStartup || seekDiscontinuity || explicitDiscontinuity) {
+            // Signal discontinuity.
+
+            if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
+                // If this was a live event this made no sense since
+                // we don't have access to all the segment before the current
+                // one.
+                mNextPTSTimeUs = getSegmentStartTimeUs(mSeqNumber);
+            }
+
+            if (seekDiscontinuity || explicitDiscontinuity) {
+                ALOGI("queueing discontinuity (seek=%d, explicit=%d)",
+                     seekDiscontinuity, explicitDiscontinuity);
 
-            queueDiscontinuity(
-                    explicitDiscontinuity
-                        ? ATSParser::DISCONTINUITY_FORMATCHANGE
-                        : ATSParser::DISCONTINUITY_SEEK,
-                    NULL /* extra */);
+                queueDiscontinuity(
+                        explicitDiscontinuity
+                            ? ATSParser::DISCONTINUITY_FORMATCHANGE
+                            : ATSParser::DISCONTINUITY_SEEK,
+                        NULL /* extra */);
+            }
         }
-    }
 
-    err = extractAndQueueAccessUnits(buffer, itemMeta);
+        err = OK;
+        if (bufferStartsWithTsSyncByte(buffer)) {
+            // Incremental extraction is only supported for MPEG2 transport streams.
+            if (tsBuffer == NULL) {
+                tsBuffer = new ABuffer(buffer->data(), buffer->capacity());
+                tsBuffer->setRange(0, 0);
+            } else if (tsBuffer->capacity() != buffer->capacity()) {
+                size_t tsOff = tsBuffer->offset(), tsSize = tsBuffer->size();
+                tsBuffer = new ABuffer(buffer->data(), buffer->capacity());
+                tsBuffer->setRange(tsOff, tsSize);
+            }
+            tsBuffer->setRange(tsBuffer->offset(), tsBuffer->size() + bytesRead);
+
+            err = extractAndQueueAccessUnitsFromTs(tsBuffer);
+        }
+
+        if (err == -EAGAIN) {
+            // bad starting sequence number hint
+            postMonitorQueue();
+            return;
+        }
+
+        if (err == ERROR_OUT_OF_RANGE) {
+            // reached stopping point
+            stopAsync(/* selfTriggered = */ true);
+            return;
+        }
+
+        if (err != OK) {
+            notifyError(err);
+            return;
+        }
+
+        mStartup = false;
+    } while (bytesRead != 0);
+
+    if (bufferStartsWithTsSyncByte(buffer)) {
+        // If we still don't see a stream after fetching a full ts segment mark it as
+        // nonexistent.
+        const size_t kNumTypes = ATSParser::NUM_SOURCE_TYPES;
+        ATSParser::SourceType srcTypes[kNumTypes] =
+                { ATSParser::VIDEO, ATSParser::AUDIO };
+        LiveSession::StreamType streamTypes[kNumTypes] =
+                { LiveSession::STREAMTYPE_VIDEO, LiveSession::STREAMTYPE_AUDIO };
+
+        for (size_t i = 0; i < kNumTypes; i++) {
+            ATSParser::SourceType srcType = srcTypes[i];
+            LiveSession::StreamType streamType = streamTypes[i];
+
+            sp<AnotherPacketSource> source =
+                static_cast<AnotherPacketSource *>(
+                    mTSParser->getSource(srcType).get());
+
+            if (source == NULL) {
+                ALOGW("MPEG2 Transport stream does not contain %s data.",
+                      srcType == ATSParser::VIDEO ? "video" : "audio");
+
+                mStreamTypeMask &= ~streamType;
+                mPacketSources.removeItem(streamType);
+            }
+        }
 
-    if (err == -EAGAIN) {
-        // bad starting sequence number hint
-        postMonitorQueue();
-        return;
     }
 
-    if (err == ERROR_OUT_OF_RANGE) {
-        // reached stopping point
-        stopAsync(/* selfTriggered = */ true);
+    if (checkDecryptPadding(buffer) != OK) {
+        ALOGE("Incorrect padding bytes after decryption.");
+        notifyError(ERROR_MALFORMED);
         return;
     }
 
+    status_t err = OK;
+    if (tsBuffer != NULL) {
+        AString method;
+        CHECK(buffer->meta()->findString("cipher-method", &method));
+        if ((tsBuffer->size() > 0 && method == "NONE")
+                || tsBuffer->size() > 16) {
+            ALOGE("MPEG2 transport stream is not an even multiple of 188 "
+                    "bytes in length.");
+            notifyError(ERROR_MALFORMED);
+            return;
+        }
+    }
+
+    // bulk extract non-ts files
+    if (tsBuffer == NULL) {
+      err = extractAndQueueAccessUnits(buffer, itemMeta);
+    }
+
     if (err != OK) {
         notifyError(err);
         return;
@@ -891,8 +992,6 @@ void PlaylistFetcher::onDownloadNext() {
     ++mSeqNumber;
 
     postMonitorQueue();
-
-    mStartup = false;
 }
 
 int32_t PlaylistFetcher::getSeqNumberForTime(int64_t timeUs) const {
@@ -927,173 +1026,163 @@ int32_t PlaylistFetcher::getSeqNumberForTime(int64_t timeUs) const {
     return firstSeqNumberInPlaylist + index;
 }
 
-status_t PlaylistFetcher::extractAndQueueAccessUnits(
-        const sp<ABuffer> &buffer, const sp<AMessage> &itemMeta) {
-    if (buffer->size() > 0 && buffer->data()[0] == 0x47) {
-        // Let's assume this is an MPEG2 transport stream.
-
-        if ((buffer->size() % 188) != 0) {
-            ALOGE("MPEG2 transport stream is not an even multiple of 188 "
-                  "bytes in length.");
-            return ERROR_MALFORMED;
-        }
-
-        if (mTSParser == NULL) {
-            // Use TS_TIMESTAMPS_ARE_ABSOLUTE so pts carry over between fetchers.
-            mTSParser = new ATSParser(ATSParser::TS_TIMESTAMPS_ARE_ABSOLUTE);
-        }
-
-        if (mNextPTSTimeUs >= 0ll) {
-            sp<AMessage> extra = new AMessage;
-            // Since we are using absolute timestamps, signal an offset of 0 to prevent
-            // ATSParser from skewing the timestamps of access units.
-            extra->setInt64(IStreamListener::kKeyMediaTimeUs, 0);
+status_t PlaylistFetcher::extractAndQueueAccessUnitsFromTs(const sp<ABuffer> &buffer) {
+    if (mTSParser == NULL) {
+        // Use TS_TIMESTAMPS_ARE_ABSOLUTE so pts carry over between fetchers.
+        mTSParser = new ATSParser(ATSParser::TS_TIMESTAMPS_ARE_ABSOLUTE);
+    }
 
-            mTSParser->signalDiscontinuity(
-                    ATSParser::DISCONTINUITY_SEEK, extra);
+    if (mNextPTSTimeUs >= 0ll) {
+        sp<AMessage> extra = new AMessage;
+        // Since we are using absolute timestamps, signal an offset of 0 to prevent
+        // ATSParser from skewing the timestamps of access units.
+        extra->setInt64(IStreamListener::kKeyMediaTimeUs, 0);
 
-            mNextPTSTimeUs = -1ll;
-        }
+        mTSParser->signalDiscontinuity(
+                ATSParser::DISCONTINUITY_SEEK, extra);
 
-        size_t offset = 0;
-        while (offset < buffer->size()) {
-            status_t err = mTSParser->feedTSPacket(buffer->data() + offset, 188);
+        mNextPTSTimeUs = -1ll;
+    }
 
-            if (err != OK) {
-                return err;
-            }
+    size_t offset = 0;
+    while (offset + 188 <= buffer->size()) {
+        status_t err = mTSParser->feedTSPacket(buffer->data() + offset, 188);
 
-            offset += 188;
+        if (err != OK) {
+            return err;
         }
 
-        status_t err = OK;
-        for (size_t i = mPacketSources.size(); i-- > 0;) {
-            sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
-
-            const char *key;
-            ATSParser::SourceType type;
-            const LiveSession::StreamType stream = mPacketSources.keyAt(i);
-            switch (stream) {
+        offset += 188;
+    }
+    // setRange to indicate consumed bytes.
+    buffer->setRange(buffer->offset() + offset, buffer->size() - offset);
 
-                case LiveSession::STREAMTYPE_VIDEO:
-                    type = ATSParser::VIDEO;
-                    key = "timeUsVideo";
-                    break;
+    status_t err = OK;
+    for (size_t i = mPacketSources.size(); i-- > 0;) {
+        sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
 
-                case LiveSession::STREAMTYPE_AUDIO:
-                    type = ATSParser::AUDIO;
-                    key = "timeUsAudio";
-                    break;
+        const char *key;
+        ATSParser::SourceType type;
+        const LiveSession::StreamType stream = mPacketSources.keyAt(i);
+        switch (stream) {
+            case LiveSession::STREAMTYPE_VIDEO:
+                type = ATSParser::VIDEO;
+                key = "timeUsVideo";
+                break;
 
-                case LiveSession::STREAMTYPE_SUBTITLES:
-                {
-                    ALOGE("MPEG2 Transport streams do not contain subtitles.");
-                    return ERROR_MALFORMED;
-                    break;
-                }
+            case LiveSession::STREAMTYPE_AUDIO:
+                type = ATSParser::AUDIO;
+                key = "timeUsAudio";
+                break;
 
-                default:
-                    TRESPASS();
+            case LiveSession::STREAMTYPE_SUBTITLES:
+            {
+                ALOGE("MPEG2 Transport streams do not contain subtitles.");
+                return ERROR_MALFORMED;
+                break;
             }
 
-            sp<AnotherPacketSource> source =
-                static_cast<AnotherPacketSource *>(
-                        mTSParser->getSource(type).get());
+            default:
+                TRESPASS();
+        }
 
-            if (source == NULL) {
-                ALOGW("MPEG2 Transport stream does not contain %s data.",
-                      type == ATSParser::VIDEO ? "video" : "audio");
+        sp<AnotherPacketSource> source =
+            static_cast<AnotherPacketSource *>(
+                    mTSParser->getSource(type).get());
 
-                mStreamTypeMask &= ~mPacketSources.keyAt(i);
-                mPacketSources.removeItemsAt(i);
-                continue;
-            }
+        if (source == NULL) {
+            continue;
+        }
 
-            int64_t timeUs;
-            sp<ABuffer> accessUnit;
-            status_t finalResult;
-            while (source->hasBufferAvailable(&finalResult)
-                    && source->dequeueAccessUnit(&accessUnit) == OK) {
-
-                CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
-                if (mMinStartTimeUs > 0) {
-                    if (timeUs < mMinStartTimeUs) {
-                        // TODO untested path
-                        // try a later ts
-                        int32_t targetDuration;
-                        mPlaylist->meta()->findInt32("target-duration", &targetDuration);
-                        int32_t incr = (mMinStartTimeUs - timeUs) / 1000000 / targetDuration;
-                        if (incr == 0) {
-                            // increment mSeqNumber by at least one
-                            incr = 1;
-                        }
-                        mSeqNumber += incr;
-                        err = -EAGAIN;
-                        break;
-                    } else {
-                        int64_t startTimeUs;
-                        if (mStartTimeUsNotify != NULL
-                                && !mStartTimeUsNotify->findInt64(key, &startTimeUs)) {
-                            mStartTimeUsNotify->setInt64(key, timeUs);
-
-                            uint32_t streamMask = 0;
-                            mStartTimeUsNotify->findInt32("streamMask", (int32_t *) &streamMask);
-                            streamMask |= mPacketSources.keyAt(i);
-                            mStartTimeUsNotify->setInt32("streamMask", streamMask);
-
-                            if (streamMask == mStreamTypeMask) {
-                                mStartTimeUsNotify->post();
-                                mStartTimeUsNotify.clear();
-                            }
+        int64_t timeUs;
+        sp<ABuffer> accessUnit;
+        status_t finalResult;
+        while (source->hasBufferAvailable(&finalResult)
+                && source->dequeueAccessUnit(&accessUnit) == OK) {
+
+            CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
+            if (mMinStartTimeUs > 0) {
+                if (timeUs < mMinStartTimeUs) {
+                    // TODO untested path
+                    // try a later ts
+                    int32_t targetDuration;
+                    mPlaylist->meta()->findInt32("target-duration", &targetDuration);
+                    int32_t incr = (mMinStartTimeUs - timeUs) / 1000000 / targetDuration;
+                    if (incr == 0) {
+                        // increment mSeqNumber by at least one
+                        incr = 1;
+                    }
+                    mSeqNumber += incr;
+                    err = -EAGAIN;
+                    break;
+                } else {
+                    int64_t startTimeUs;
+                    if (mStartTimeUsNotify != NULL
+                            && !mStartTimeUsNotify->findInt64(key, &startTimeUs)) {
+                        mStartTimeUsNotify->setInt64(key, timeUs);
+
+                        uint32_t streamMask = 0;
+                        mStartTimeUsNotify->findInt32("streamMask", (int32_t *) &streamMask);
+                        streamMask |= mPacketSources.keyAt(i);
+                        mStartTimeUsNotify->setInt32("streamMask", streamMask);
+
+                        if (streamMask == mStreamTypeMask) {
+                            mStartTimeUsNotify->post();
+                            mStartTimeUsNotify.clear();
                         }
                     }
                 }
+            }
 
-                if (mStopParams != NULL) {
-                    // Queue discontinuity in original stream.
-                    int64_t stopTimeUs;
-                    if (!mStopParams->findInt64(key, &stopTimeUs) || timeUs >= stopTimeUs) {
-                        packetSource->queueAccessUnit(mSession->createFormatChangeBuffer());
-                        mStreamTypeMask &= ~stream;
-                        mPacketSources.removeItemsAt(i);
-                        break;
-                    }
+            if (mStopParams != NULL) {
+                // Queue discontinuity in original stream.
+                int64_t stopTimeUs;
+                if (!mStopParams->findInt64(key, &stopTimeUs) || timeUs >= stopTimeUs) {
+                    packetSource->queueAccessUnit(mSession->createFormatChangeBuffer());
+                    mStreamTypeMask &= ~stream;
+                    mPacketSources.removeItemsAt(i);
+                    break;
                 }
+            }
 
-                // Note that we do NOT dequeue any discontinuities except for format change.
-
-                // for simplicity, store a reference to the format in each unit
-                sp<MetaData> format = source->getFormat();
-                if (format != NULL) {
-                    accessUnit->meta()->setObject("format", format);
-                }
+            // Note that we do NOT dequeue any discontinuities except for format change.
 
-                // Stash the sequence number so we can hint future fetchers where to start at.
-                accessUnit->meta()->setInt32("seq", mSeqNumber);
-                packetSource->queueAccessUnit(accessUnit);
+            // for simplicity, store a reference to the format in each unit
+            sp<MetaData> format = source->getFormat();
+            if (format != NULL) {
+                accessUnit->meta()->setObject("format", format);
             }
 
-            if (err != OK) {
-                break;
-            }
+            // Stash the sequence number so we can hint future playlist where to start at.
+            accessUnit->meta()->setInt32("seq", mSeqNumber);
+            packetSource->queueAccessUnit(accessUnit);
         }
 
         if (err != OK) {
-            for (size_t i = mPacketSources.size(); i-- > 0;) {
-                sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
-                packetSource->clear();
-            }
-            return err;
+            break;
         }
+    }
 
-        if (!mStreamTypeMask) {
-            // Signal gap is filled between original and new stream.
-            ALOGV("ERROR OUT OF RANGE");
-            return ERROR_OUT_OF_RANGE;
+    if (err != OK) {
+        for (size_t i = mPacketSources.size(); i-- > 0;) {
+            sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
+            packetSource->clear();
         }
+        return err;
+    }
 
-        return OK;
-    } else if (buffer->size() >= 7 && !memcmp("WEBVTT\n", buffer->data(), 7)) {
+    if (!mStreamTypeMask) {
+        // Signal gap is filled between original and new stream.
+        ALOGV("ERROR OUT OF RANGE");
+        return ERROR_OUT_OF_RANGE;
+    }
+
+    return OK;
+}
+
+status_t PlaylistFetcher::extractAndQueueAccessUnits(
+        const sp<ABuffer> &buffer, const sp<AMessage> &itemMeta) {
+    if (buffer->size() >= 7 && !memcmp("WEBVTT\n", buffer->data(), 7)) {
         if (mStreamTypeMask != LiveSession::STREAMTYPE_SUBTITLES) {
             ALOGE("This stream only contains subtitles.");
             return ERROR_MALFORMED;
diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h
index 8404b8d..7e21523 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.h
+++ b/media/libstagefright/httplive/PlaylistFetcher.h
@@ -87,8 +87,11 @@ private:
 
     static const int64_t kMinBufferedDurationUs;
     static const int64_t kMaxMonitorDelayUs;
+    static const int32_t kDownloadBlockSize;
     static const int32_t kNumSkipFrames;
 
+    static bool bufferStartsWithTsSyncByte(const sp<ABuffer>& buffer);
+
     // notifications to mSession
     sp<AMessage> mNotify;
     sp<AMessage> mStartTimeUsNotify;
@@ -169,6 +172,8 @@ private:
     // Resume a fetcher to continue until the stopping point stored in msg.
     status_t onResumeUntil(const sp<AMessage> &msg);
 
+    status_t extractAndQueueAccessUnitsFromTs(const sp<ABuffer> &buffer);
+
     status_t extractAndQueueAccessUnits(
             const sp<ABuffer> &buffer, const sp<AMessage> &itemMeta);
 
diff --git a/media/libstagefright/mpeg2ts/ATSParser.h b/media/libstagefright/mpeg2ts/ATSParser.h
index a10edc9..8a80069 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.h
+++ b/media/libstagefright/mpeg2ts/ATSParser.h
@@ -71,8 +71,9 @@ struct ATSParser : public RefBase {
     void signalEOS(status_t finalResult);
 
     enum SourceType {
-        VIDEO,
-        AUDIO
+        VIDEO = 0,
+        AUDIO = 1,
+        NUM_SOURCE_TYPES = 2
     };
     sp<MediaSource> getSource(SourceType type);
 
-- 
cgit v1.1


From b635b0e66b257ab442e230bca96afd5105cf6829 Mon Sep 17 00:00:00 2001
From: Lajos Molnar <lajos@google.com>
Date: Thu, 13 Feb 2014 15:29:49 -0800
Subject: mediaplayer: keep more buffers with the BufferQueue

Change OMX buffer allocation policy to allocate
nBufferCountMin + what is required for the BQ.

For the BQ, try to allocate 2 additional buffers than
the minimum undequeued count.

Also account for the fact that BQ may return one less
than the actual minimum undequeued count.

In most cases the resulting number of buffers ends up
being the same as with the previous policy, but we
keep more buffers with the BQ.

Change-Id: I826db8bf7dd333b620299dba60bf1b81b228275d
Bug: 13170236
---
 media/libstagefright/ACodec.cpp   | 47 ++++++++++++++++++++++-----------------
 media/libstagefright/OMXCodec.cpp | 38 +++++++++++++++++++++++--------
 2 files changed, 55 insertions(+), 30 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 9276818..14d99cf 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -638,18 +638,33 @@ status_t ACodec::configureOutputBuffersFromNativeWindow(
         return err;
     }
 
-    // XXX: Is this the right logic to use?  It's not clear to me what the OMX
-    // buffer counts refer to - how do they account for the renderer holding on
-    // to buffers?
-    if (def.nBufferCountActual < def.nBufferCountMin + *minUndequeuedBuffers) {
-        OMX_U32 newBufferCount = def.nBufferCountMin + *minUndequeuedBuffers;
+    // FIXME: assume that surface is controlled by app (native window
+    // returns the number for the case when surface is not controlled by app)
+    (*minUndequeuedBuffers)++;
+
+
+    // Use conservative allocation while also trying to reduce starvation
+    //
+    // 1. allocate at least nBufferCountMin + minUndequeuedBuffers - that is the
+    //    minimum needed for the consumer to be able to work
+    // 2. try to allocate two (2) additional buffers to reduce starvation from
+    //    the consumer
+    for (OMX_U32 extraBuffers = 2; /* condition inside loop */; extraBuffers--) {
+        OMX_U32 newBufferCount =
+            def.nBufferCountMin + *minUndequeuedBuffers + extraBuffers;
         def.nBufferCountActual = newBufferCount;
         err = mOMX->setParameter(
                 mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
 
-        if (err != OK) {
-            ALOGE("[%s] setting nBufferCountActual to %lu failed: %d",
-                    mComponentName.c_str(), newBufferCount, err);
+        if (err == OK) {
+            *minUndequeuedBuffers += extraBuffers;
+            break;
+        }
+
+        ALOGW("[%s] setting nBufferCountActual to %lu failed: %d",
+                mComponentName.c_str(), newBufferCount, err);
+        /* exit condition */
+        if (extraBuffers == 0) {
             return err;
         }
     }
@@ -674,6 +689,7 @@ status_t ACodec::allocateOutputBuffersFromNativeWindow() {
             &bufferCount, &bufferSize, &minUndequeuedBuffers);
     if (err != 0)
         return err;
+    mNumUndequeuedBuffers = minUndequeuedBuffers;
 
     ALOGV("[%s] Allocating %lu buffers from a native window of size %lu on "
          "output port",
@@ -739,6 +755,7 @@ status_t ACodec::allocateOutputMetaDataBuffers() {
             &bufferCount, &bufferSize, &minUndequeuedBuffers);
     if (err != 0)
         return err;
+    mNumUndequeuedBuffers = minUndequeuedBuffers;
 
     ALOGV("[%s] Allocating %lu meta buffers on output port",
          mComponentName.c_str(), bufferCount);
@@ -2429,19 +2446,7 @@ void ACodec::waitUntilAllPossibleNativeWindowBuffersAreReturnedToUs() {
         return;
     }
 
-    int minUndequeuedBufs = 0;
-    status_t err = mNativeWindow->query(
-            mNativeWindow.get(), NATIVE_WINDOW_MIN_UNDEQUEUED_BUFFERS,
-            &minUndequeuedBufs);
-
-    if (err != OK) {
-        ALOGE("[%s] NATIVE_WINDOW_MIN_UNDEQUEUED_BUFFERS query failed: %s (%d)",
-                mComponentName.c_str(), strerror(-err), -err);
-
-        minUndequeuedBufs = 0;
-    }
-
-    while (countBuffersOwnedByNativeWindow() > (size_t)minUndequeuedBufs
+    while (countBuffersOwnedByNativeWindow() > mNumUndequeuedBuffers
             && dequeueBufferFromNativeWindow() != NULL) {
         // these buffers will be submitted as regular buffers; account for this
         if (mStoreMetaDataInOutputBuffers && mMetaDataBuffersToSubmit > 0) {
diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp
index 43736ad..79a9665 100644
--- a/media/libstagefright/OMXCodec.cpp
+++ b/media/libstagefright/OMXCodec.cpp
@@ -92,6 +92,7 @@ static sp<MediaSource> InstantiateSoftwareEncoder(
 
 #define CODEC_LOGI(x, ...) ALOGI("[%s] "x, mComponentName, ##__VA_ARGS__)
 #define CODEC_LOGV(x, ...) ALOGV("[%s] "x, mComponentName, ##__VA_ARGS__)
+#define CODEC_LOGW(x, ...) ALOGW("[%s] "x, mComponentName, ##__VA_ARGS__)
 #define CODEC_LOGE(x, ...) ALOGE("[%s] "x, mComponentName, ##__VA_ARGS__)
 
 struct OMXCodecObserver : public BnOMXObserver {
@@ -1777,21 +1778,40 @@ status_t OMXCodec::allocateOutputBuffersFromNativeWindow() {
                 strerror(-err), -err);
         return err;
     }
-
-    // XXX: Is this the right logic to use?  It's not clear to me what the OMX
-    // buffer counts refer to - how do they account for the renderer holding on
-    // to buffers?
-    if (def.nBufferCountActual < def.nBufferCountMin + minUndequeuedBufs) {
-        OMX_U32 newBufferCount = def.nBufferCountMin + minUndequeuedBufs;
+    // FIXME: assume that surface is controlled by app (native window
+    // returns the number for the case when surface is not controlled by app)
+    minUndequeuedBufs++;
+
+    // Use conservative allocation while also trying to reduce starvation
+    //
+    // 1. allocate at least nBufferCountMin + minUndequeuedBuffers - that is the
+    //    minimum needed for the consumer to be able to work
+    // 2. try to allocate two (2) additional buffers to reduce starvation from
+    //    the consumer
+    CODEC_LOGI("OMX-buffers: min=%u actual=%u undeq=%d",
+            def.nBufferCountMin, def.nBufferCountActual, minUndequeuedBufs);
+
+    for (OMX_U32 extraBuffers = 2; /* condition inside loop */; extraBuffers--) {
+        OMX_U32 newBufferCount =
+            def.nBufferCountMin + minUndequeuedBufs + extraBuffers;
         def.nBufferCountActual = newBufferCount;
         err = mOMX->setParameter(
                 mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
-        if (err != OK) {
-            CODEC_LOGE("setting nBufferCountActual to %lu failed: %d",
-                    newBufferCount, err);
+
+        if (err == OK) {
+            minUndequeuedBufs += extraBuffers;
+            break;
+        }
+
+        CODEC_LOGW("setting nBufferCountActual to %lu failed: %d",
+                newBufferCount, err);
+        /* exit condition */
+        if (extraBuffers == 0) {
             return err;
         }
     }
+    CODEC_LOGI("OMX-buffers: min=%u actual=%u undeq=%d",
+            def.nBufferCountMin, def.nBufferCountActual, minUndequeuedBufs);
 
     err = native_window_set_buffer_count(
             mNativeWindow.get(), def.nBufferCountActual);
-- 
cgit v1.1


From 0c271b750f7314e187f8479f478ad9382d9a9045 Mon Sep 17 00:00:00 2001
From: Marco Nelissen <marcone@google.com>
Date: Thu, 13 Mar 2014 14:18:25 -0700
Subject: Verify certificates

b/13418320

Cherrypicked from 8a57d24c733b08da846b54d1adf029e606b5a5f3
https://googleplex-android-review.git.corp.google.com/#/c/433309/

Change-Id: I3052dd5f9ec057e700784cd713f6a7dab9ecfe7b
---
 media/libstagefright/chromium_http/support.cpp | 104 ++++++++++++++++++++++++-
 1 file changed, 102 insertions(+), 2 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/chromium_http/support.cpp b/media/libstagefright/chromium_http/support.cpp
index 13ae3df..0ecf8cc 100644
--- a/media/libstagefright/chromium_http/support.cpp
+++ b/media/libstagefright/chromium_http/support.cpp
@@ -34,14 +34,97 @@
 #include "net/proxy/proxy_config_service_android.h"
 
 #include "include/ChromiumHTTPDataSource.h"
-
+#include <arpa/inet.h>
+#include <binder/Parcel.h>
 #include <cutils/log.h>
 #include <cutils/properties.h>
 #include <media/stagefright/MediaErrors.h>
 #include <string>
 
+#include <utils/Errors.h>
+#include <binder/IInterface.h>
+#include <binder/IServiceManager.h>
+
 namespace android {
 
+// must be kept in sync with interface defined in IAudioService.aidl
+class IAudioService : public IInterface
+{
+public:
+    DECLARE_META_INTERFACE(AudioService);
+
+    virtual int verifyX509CertChain(
+            const std::vector<std::string>& cert_chain,
+            const std::string& hostname,
+            const std::string& auth_type) = 0;
+};
+
+class BpAudioService : public BpInterface<IAudioService>
+{
+public:
+    BpAudioService(const sp<IBinder>& impl)
+        : BpInterface<IAudioService>(impl)
+    {
+    }
+
+    virtual int verifyX509CertChain(
+            const std::vector<std::string>& cert_chain,
+            const std::string& hostname,
+            const std::string& auth_type)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioService::getInterfaceDescriptor());
+
+        // The vector of std::string we get isn't really a vector of strings,
+        // but rather a vector of binary certificate data. If we try to pass
+        // it to Java language code as a string, it ends up mangled on the other
+        // side, so send them as bytes instead.
+        // Since we can't send an array of byte arrays, send a single array,
+        // which will be split out by the recipient.
+
+        int numcerts = cert_chain.size();
+        data.writeInt32(numcerts);
+        size_t total = 0;
+        for (int i = 0; i < numcerts; i++) {
+            total += cert_chain[i].size();
+        }
+        size_t bytesize = total + numcerts * 4;
+        uint8_t *bytes = (uint8_t*) malloc(bytesize);
+        if (!bytes) {
+            return 5; // SSL_INVALID
+        }
+        ALOGV("%d certs: %d -> %d", numcerts, total, bytesize);
+
+        int offset = 0;
+        for (int i = 0; i < numcerts; i++) {
+            int32_t certsize = cert_chain[i].size();
+            // store this in a known order, which just happens to match the default
+            // byte order of a java ByteBuffer
+            int32_t bigsize = htonl(certsize);
+            ALOGV("cert %d, size %d", i, certsize);
+            memcpy(bytes + offset, &bigsize, sizeof(bigsize));
+            offset += sizeof(bigsize);
+            memcpy(bytes + offset, cert_chain[i].data(), certsize);
+            offset += certsize;
+        }
+        data.writeByteArray(bytesize, bytes);
+        free(bytes);
+        data.writeString16(String16(hostname.c_str()));
+        data.writeString16(String16(auth_type.c_str()));
+
+        int32_t result;
+        if (remote()->transact(IBinder::FIRST_CALL_TRANSACTION, data, &reply) != NO_ERROR
+                || reply.readExceptionCode() < 0 || reply.readInt32(&result) != NO_ERROR) {
+            return 5; // SSL_INVALID;
+        }
+        return result;
+    }
+
+};
+
+IMPLEMENT_META_INTERFACE(AudioService, "android.media.IAudioService");
+
+
 static Mutex gNetworkThreadLock;
 static base::Thread *gNetworkThread = NULL;
 static scoped_refptr<net::URLRequestContext> gReqContext;
@@ -211,7 +294,24 @@ SfNetworkLibrary::VerifyResult SfNetworkLibrary::VerifyX509CertChain(
         const std::vector<std::string>& cert_chain,
         const std::string& hostname,
         const std::string& auth_type) {
-    return VERIFY_OK;
+
+    sp<IBinder> binder =
+        defaultServiceManager()->checkService(String16("audio"));
+    if (binder == 0) {
+        ALOGW("Thread cannot connect to the audio service");
+    } else {
+        sp<IAudioService> service = interface_cast<IAudioService>(binder);
+        int code = service->verifyX509CertChain(cert_chain, hostname, auth_type);
+        ALOGV("verified: %d", code);
+        if (code == -1) {
+            return VERIFY_OK;
+        } else if (code == 2) { // SSL_IDMISMATCH
+            return VERIFY_BAD_HOSTNAME;
+        } else if (code == 3) { // SSL_UNTRUSTED
+            return VERIFY_NO_TRUSTED_ROOT;
+        }
+    }
+    return VERIFY_INVOCATION_ERROR;
 }
 
 ////////////////////////////////////////////////////////////////////////////////
-- 
cgit v1.1


From 1faa41704e0b976e546321effcb09a85767d51ba Mon Sep 17 00:00:00 2001
From: Lajos Molnar <lajos@google.com>
Date: Wed, 19 Mar 2014 12:22:01 -0700
Subject: stagefright: allow for minUndequeuedBufs to be one less

Bug: 13533477
Change-Id: I909324e3e24f65d8051e0d5474267620efcfc729
---
 media/libstagefright/ACodec.cpp   |  7 ++++---
 media/libstagefright/OMXCodec.cpp | 10 ++++++----
 2 files changed, 10 insertions(+), 7 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 14d99cf..3ad3ca7 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -640,8 +640,8 @@ status_t ACodec::configureOutputBuffersFromNativeWindow(
 
     // FIXME: assume that surface is controlled by app (native window
     // returns the number for the case when surface is not controlled by app)
-    (*minUndequeuedBuffers)++;
-
+    // FIXME2: This means that minUndeqeueudBufs can be 1 larger than reported
+    // For now, try to allocate 1 more buffer, but don't fail if unsuccessful
 
     // Use conservative allocation while also trying to reduce starvation
     //
@@ -649,7 +649,8 @@ status_t ACodec::configureOutputBuffersFromNativeWindow(
     //    minimum needed for the consumer to be able to work
     // 2. try to allocate two (2) additional buffers to reduce starvation from
     //    the consumer
-    for (OMX_U32 extraBuffers = 2; /* condition inside loop */; extraBuffers--) {
+    //    plus an extra buffer to account for incorrect minUndequeuedBufs
+    for (OMX_U32 extraBuffers = 2 + 1; /* condition inside loop */; extraBuffers--) {
         OMX_U32 newBufferCount =
             def.nBufferCountMin + *minUndequeuedBuffers + extraBuffers;
         def.nBufferCountActual = newBufferCount;
diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp
index 79a9665..070e438 100644
--- a/media/libstagefright/OMXCodec.cpp
+++ b/media/libstagefright/OMXCodec.cpp
@@ -1780,7 +1780,8 @@ status_t OMXCodec::allocateOutputBuffersFromNativeWindow() {
     }
     // FIXME: assume that surface is controlled by app (native window
     // returns the number for the case when surface is not controlled by app)
-    minUndequeuedBufs++;
+    // FIXME2: This means that minUndeqeueudBufs can be 1 larger than reported
+    // For now, try to allocate 1 more buffer, but don't fail if unsuccessful
 
     // Use conservative allocation while also trying to reduce starvation
     //
@@ -1788,10 +1789,11 @@ status_t OMXCodec::allocateOutputBuffersFromNativeWindow() {
     //    minimum needed for the consumer to be able to work
     // 2. try to allocate two (2) additional buffers to reduce starvation from
     //    the consumer
-    CODEC_LOGI("OMX-buffers: min=%u actual=%u undeq=%d",
+    //    plus an extra buffer to account for incorrect minUndequeuedBufs
+    CODEC_LOGI("OMX-buffers: min=%u actual=%u undeq=%d+1",
             def.nBufferCountMin, def.nBufferCountActual, minUndequeuedBufs);
 
-    for (OMX_U32 extraBuffers = 2; /* condition inside loop */; extraBuffers--) {
+    for (OMX_U32 extraBuffers = 2 + 1; /* condition inside loop */; extraBuffers--) {
         OMX_U32 newBufferCount =
             def.nBufferCountMin + minUndequeuedBufs + extraBuffers;
         def.nBufferCountActual = newBufferCount;
@@ -1810,7 +1812,7 @@ status_t OMXCodec::allocateOutputBuffersFromNativeWindow() {
             return err;
         }
     }
-    CODEC_LOGI("OMX-buffers: min=%u actual=%u undeq=%d",
+    CODEC_LOGI("OMX-buffers: min=%u actual=%u undeq=%d+1",
             def.nBufferCountMin, def.nBufferCountActual, minUndequeuedBufs);
 
     err = native_window_set_buffer_count(
-- 
cgit v1.1


From 25bab830af062dcaef6f75220066b221fd30957e Mon Sep 17 00:00:00 2001
From: Marco Nelissen <marcone@google.com>
Date: Thu, 20 Mar 2014 12:32:48 -0700
Subject: Fix build

As of jb-mr1, need to explicitly include library dependencies. The original
change in jb, which the automerger dutyfully merged, did not need this.

Change-Id: I429637e2d01b8084376bb97a02bb091a0ea9eaf7
---
 media/libstagefright/chromium_http/Android.mk | 1 +
 1 file changed, 1 insertion(+)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/chromium_http/Android.mk b/media/libstagefright/chromium_http/Android.mk
index 2c6d84c..75c2bed 100644
--- a/media/libstagefright/chromium_http/Android.mk
+++ b/media/libstagefright/chromium_http/Android.mk
@@ -18,6 +18,7 @@ LOCAL_C_INCLUDES:= \
 LOCAL_CFLAGS += -Wno-multichar
 
 LOCAL_SHARED_LIBRARIES += \
+        libbinder \
         libstlport \
         libchromium_net \
         libutils \
-- 
cgit v1.1


From f622e2c78407b50806a53a06a89b2b6546420f55 Mon Sep 17 00:00:00 2001
From: Rachad <rachad@google.com>
Date: Mon, 10 Feb 2014 16:54:46 -0800
Subject: Increased recorded MP4 max file size to 4GB (more precisely 2^32-1
 bytes -- FAT32 max file size)

bug: 11039801
Change-Id: I6169f798c59d123d02d5fd7afa3b9e645ebdb598
(cherry picked from commit 1f1f2b1678fd0d038dfc501252dd2b65ecf10cae)
---
 media/libstagefright/MPEG4Writer.cpp | 8 +++++---
 1 file changed, 5 insertions(+), 3 deletions(-)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index a0f17b5..42885dd 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -43,7 +43,9 @@
 namespace android {
 
 static const int64_t kMinStreamableFileSizeInBytes = 5 * 1024 * 1024;
-static const int64_t kMax32BitFileSize = 0x007fffffffLL;
+static const int64_t kMax32BitFileSize = 0x00ffffffffLL; // 2^32-1 : max FAT32
+                                                         // filesystem file size
+                                                         // used by most SD cards
 static const uint8_t kNalUnitTypeSeqParamSet = 0x07;
 static const uint8_t kNalUnitTypePicParamSet = 0x08;
 static const int64_t kInitialDelayTimeUs     = 700000LL;
@@ -860,11 +862,11 @@ status_t MPEG4Writer::reset() {
     // Fix up the size of the 'mdat' chunk.
     if (mUse32BitOffset) {
         lseek64(mFd, mMdatOffset, SEEK_SET);
-        int32_t size = htonl(static_cast<int32_t>(mOffset - mMdatOffset));
+        uint32_t size = htonl(static_cast<uint32_t>(mOffset - mMdatOffset));
         ::write(mFd, &size, 4);
     } else {
         lseek64(mFd, mMdatOffset + 8, SEEK_SET);
-        int64_t size = mOffset - mMdatOffset;
+        uint64_t size = mOffset - mMdatOffset;
         size = hton64(size);
         ::write(mFd, &size, 8);
     }
-- 
cgit v1.1


From 65e20ffc984c541a8119420f917493dd7b703f77 Mon Sep 17 00:00:00 2001
From: Robert Shih <robertshih@google.com>
Date: Fri, 7 Feb 2014 12:26:58 -0800
Subject: DO NOT MERGE: PlaylistFetcher: fix infinite loop when parsing ADTS.

First check for embedded ID3 tag, then bail out if invalid.

Bug: 12934795
Change-Id: I74acebed4bfb2c6ca44dfe936166fdba8510233f
---
 media/libstagefright/httplive/PlaylistFetcher.cpp | 12 ++++++++++++
 1 file changed, 12 insertions(+)

(limited to 'media/libstagefright')

diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index ada856d..668cbd4 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -1321,6 +1321,18 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits(
             | (adtsHeader[4] << 3)
             | (adtsHeader[5] >> 5);
 
+        if (aac_frame_length == 0) {
+            const uint8_t *id3Header = adtsHeader;
+            if (!memcmp(id3Header, "ID3", 3)) {
+                ID3 id3(id3Header, buffer->size() - offset, true);
+                if (id3.isValid()) {
+                    offset += id3.rawSize();
+                    continue;
+                };
+            }
+            return ERROR_MALFORMED;
+        }
+
         CHECK_LE(offset + aac_frame_length, buffer->size());
 
         sp<ABuffer> unit = new ABuffer(aac_frame_length);
-- 
cgit v1.1