(Auto)update libjingle 67686255-> 67689476

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6216 4adac7df-926f-26a2-2b94-8c16560cd09d

talk/base/timeutils.cc

@@ -203,4 +203,18 @@ int32 TimeDiff(uint32 later, uint32 earlier) {
 #endif
 }
 
+TimestampWrapAroundHandler::TimestampWrapAroundHandler()
+    : last_ts_(0), num_wrap_(0) {}
+
+int64 TimestampWrapAroundHandler::Unwrap(uint32 ts) {
+  if (ts < last_ts_) {
+    if (last_ts_ > 0xf0000000 && ts < 0x0fffffff) {
+      ++num_wrap_;
+    }
+  }
+  last_ts_ = ts;
+  int64_t unwrapped_ts = ts + (num_wrap_ << 32);
+  return unwrapped_ts;
+}
+
 } // namespace talk_base

talk/base/timeutils.h

@@ -97,6 +97,17 @@ inline int64 UnixTimestampNanosecsToNtpMillisecs(int64 unix_ts_ns) {
   return unix_ts_ns / kNumNanosecsPerMillisec + kJan1970AsNtpMillisecs;
 }
 
+class TimestampWrapAroundHandler {
+ public:
+  TimestampWrapAroundHandler();
+
+  int64 Unwrap(uint32 ts);
+
+ private:
+  uint32 last_ts_;
+  int64 num_wrap_;
+};
+
 }  // namespace talk_base
 
 #endif  // TALK_BASE_TIMEUTILS_H_

talk/base/timeutils_unittest.cc

@@ -160,4 +160,27 @@ TEST(TimeTest, DISABLED_CurrentTmTime) {
   EXPECT_TRUE(0 <= microseconds && microseconds < 1000000);
 }
 
+class TimestampWrapAroundHandlerTest : public testing::Test {
+ public:
+  TimestampWrapAroundHandlerTest() {}
+
+ protected:
+  TimestampWrapAroundHandler wraparound_handler_;
+};
+
+TEST_F(TimestampWrapAroundHandlerTest, Unwrap) {
+  uint32 ts = 0xfffffff2;
+  int64 unwrapped_ts = ts;
+  EXPECT_EQ(ts, wraparound_handler_.Unwrap(ts));
+  ts = 2;
+  unwrapped_ts += 0x10;
+  EXPECT_EQ(unwrapped_ts, wraparound_handler_.Unwrap(ts));
+  ts = 0xfffffff2;
+  unwrapped_ts += 0xfffffff0;
+  EXPECT_EQ(unwrapped_ts, wraparound_handler_.Unwrap(ts));
+  ts = 0;
+  unwrapped_ts += 0xe;
+  EXPECT_EQ(unwrapped_ts, wraparound_handler_.Unwrap(ts));
+}
+
 }  // namespace talk_base

talk/media/webrtc/webrtcvideoengine.cc

@@ -180,8 +180,7 @@ class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
         channel_id_(channel_id),
         width_(0),
         height_(0),
-        first_frame_arrived_(false),
+        capture_start_rtp_time_stamp_(-1),
-        capture_start_rtp_time_stamp_(0),
         capture_start_ntp_time_ms_(0) {
   }
 
@@ -233,8 +232,7 @@ class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
                            int64_t render_time,
                            void* handle) {
     talk_base::CritScope cs(&crit_);
-    if (!first_frame_arrived_) {
+    if (capture_start_rtp_time_stamp_ < 0) {
-      first_frame_arrived_ = true;
       capture_start_rtp_time_stamp_ = rtp_time_stamp;
     }
 
@@ -242,9 +240,9 @@ class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
 
 #ifdef USE_WEBRTC_DEV_BRANCH
     if (ntp_time_ms > 0) {
-      uint32 elapsed_time_ms =
+      int64 elapsed_time_ms =
-          (rtp_time_stamp - capture_start_rtp_time_stamp_) /
+          (rtp_ts_wraparound_handler_.Unwrap(rtp_time_stamp) -
-          kVideoCodecClockratekHz;
+           capture_start_rtp_time_stamp_) / kVideoCodecClockratekHz;
       capture_start_ntp_time_ms_ = ntp_time_ms - elapsed_time_ms;
     }
 #endif
@@ -329,8 +327,8 @@ class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
   unsigned int width_;
   unsigned int height_;
   talk_base::RateTracker frame_rate_tracker_;
-  bool first_frame_arrived_;
+  talk_base::TimestampWrapAroundHandler rtp_ts_wraparound_handler_;
-  uint32 capture_start_rtp_time_stamp_;
+  int64 capture_start_rtp_time_stamp_;
   int64 capture_start_ntp_time_ms_;
 };