commit 1508809cf9b1fba562566ae0372e4f3a1a840731
author Wei Jia <wjia@google.com> Fri Jun 03 02:08:50 2016 +0000
committer android-build-merger <android-build-merger@google.com> Fri Jun 03 02:08:50 2016 +0000
tree 4026df3ca6f58af6c2b07d603b5e03691dae220e
parent d2958597bc86d7eab85f007997d8986e1542f7a4
parent 4cf0436f204a4f67debfb7c0022cf7100232c8db

Merge "RTSP: add timeout scheme to handle error cases." into nyc-dev
am: 4cf0436f20

* commit '4cf0436f204a4f67debfb7c0022cf7100232c8db':
  RTSP: add timeout scheme to handle error cases.

Change-Id: I4f5e7d5a25b447577092d898a3e9cc629a5a2fd2

media/libstagefright/rtsp/MyHandler.h

diff --git a/media/libstagefright/rtsp/MyHandler.h b/media/libstagefright/rtsp/MyHandler.h
index bef9c0a..f9a9ab9 100644
--- a/media/libstagefright/rtsp/MyHandler.h
+++ b/media/libstagefright/rtsp/MyHandler.h
@@ -64,6 +64,10 @@
 
 static int64_t kPauseDelayUs = 3000000ll;
 
+// The allowed maximum number of stale access units at the beginning of
+// a new sequence.
+static int32_t kMaxAllowedStaleAccessUnits = 20;
+
 namespace android {
 
 static bool GetAttribute(const char *s, const char *key, AString *value) {
@@ -1049,14 +1053,24 @@
                 }
 
                 if (track->mNewSegment) {
-                    // The sequence number from RTP packet has only 16 bits and is adjusted
-                    // by the client. Only the low 16 bits of seq in RTP-Info of reply of
-                    // RTSP "PLAY" command should be used to detect the first RTP patcket
+                    // The sequence number from RTP packet has only 16 bits and is extended
+                    // by ARTPSource. Only the low 16 bits of seq in RTP-Info of reply of
+                    // RTSP "PLAY" command should be used to detect the first RTP packet
                     // after seeking.
-                    if ((((seqNum ^ track->mFirstSeqNumInSegment) & 0xffff) != 0)) {
-                        // Not the first rtp packet of the stream after seeking, discarding.
-                        ALOGV("discarding stale access unit (0x%x : 0x%x)",
-                             seqNum, track->mFirstSeqNumInSegment);
+                    if (track->mAllowedStaleAccessUnits > 0) {
+                        if ((((seqNum ^ track->mFirstSeqNumInSegment) & 0xffff) != 0)) {
+                            // Not the first rtp packet of the stream after seeking, discarding.
+                            track->mAllowedStaleAccessUnits--;
+                            ALOGV("discarding stale access unit (0x%x : 0x%x)",
+                                 seqNum, track->mFirstSeqNumInSegment);
+                            break;
+                        }
+                    } else { // track->mAllowedStaleAccessUnits <= 0
+                        mNumAccessUnitsReceived = 0;
+                        ALOGW_IF(track->mAllowedStaleAccessUnits == 0,
+                             "Still no first rtp packet after %d stale ones",
+                             kMaxAllowedStaleAccessUnits);
+                        track->mAllowedStaleAccessUnits = -1;
                         break;
                     }
 
@@ -1516,6 +1530,7 @@
             TrackInfo *info = &mTracks.editItemAt(trackIndex);
             info->mFirstSeqNumInSegment = seq;
             info->mNewSegment = true;
+            info->mAllowedStaleAccessUnits = kMaxAllowedStaleAccessUnits;
 
             CHECK(GetAttribute((*it).c_str(), "rtptime", &val));
 
@@ -1559,6 +1574,7 @@
         bool mUsingInterleavedTCP;
         uint32_t mFirstSeqNumInSegment;
         bool mNewSegment;
+        int32_t mAllowedStaleAccessUnits;
 
         uint32_t mRTPAnchor;
         int64_t mNTPAnchorUs;
@@ -1642,6 +1658,7 @@
         info->mUsingInterleavedTCP = false;
         info->mFirstSeqNumInSegment = 0;
         info->mNewSegment = true;
+        info->mAllowedStaleAccessUnits = kMaxAllowedStaleAccessUnits;
         info->mRTPSocket = -1;
         info->mRTCPSocket = -1;
         info->mRTPAnchor = 0;