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authorandrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>2013-10-22 12:50:00 +0000
committerandrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>2013-10-22 12:50:00 +0000
commit221798a7c4ad83ba7fb8b3ae39b99d44278bbe5a (patch)
treec496e823364f8d8bffaa37bdba1b9995614d1e58 /modules/video_coding
parentb45ab8b2e985f1429e5a2ed27c798f887f0c5fc9 (diff)
downloadwebrtc-221798a7c4ad83ba7fb8b3ae39b99d44278bbe5a.tar.gz
Upgrade scoped_ptr to Chromium's latest version.
Analogous to the recent libjingle change: http://cl/54929753-p10. This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather than scoped_array and scoped_ptr_malloc respectively. - Add Chromium's template-based COMPILE_ASSERT. We didn't have this previously in order to support the macro in C. Instead, move the existing macro to compile_assert_c.h. - Additionally copy the move.h and template_util.h depedencies and add the WARN_UNUSED_RESULT macro. - Leave scoped_array and scoped_ptr_malloc for now, but mark as deprecated. - Remove scoped_ptr foo(NULL) use. The default constructor handles it. - Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc. - Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove some repeated code. TESTED=trybots R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2449005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5015 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'modules/video_coding')
-rw-r--r--modules/video_coding/main/test/rtp_player.cc1
1 files changed, 0 insertions, 1 deletions
diff --git a/modules/video_coding/main/test/rtp_player.cc b/modules/video_coding/main/test/rtp_player.cc
index 6af43893..99651431 100644
--- a/modules/video_coding/main/test/rtp_player.cc
+++ b/modules/video_coding/main/test/rtp_player.cc
@@ -326,7 +326,6 @@ class RtpPlayerImpl : public RtpPlayerInterface {
float loss_rate, uint32_t rtt_ms, bool reordering)
: ssrc_handlers_(payload_sink_factory, payload_types),
clock_(clock),
- packet_source_(NULL),
next_rtp_time_(0),
first_packet_(true),
first_packet_rtp_time_(0),