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author | andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2013-10-22 12:50:00 +0000 |
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committer | andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2013-10-22 12:50:00 +0000 |
commit | 221798a7c4ad83ba7fb8b3ae39b99d44278bbe5a (patch) | |
tree | c496e823364f8d8bffaa37bdba1b9995614d1e58 /modules/video_coding | |
parent | b45ab8b2e985f1429e5a2ed27c798f887f0c5fc9 (diff) | |
download | webrtc-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.cc | 1 |
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), |