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authorBo Liu <boliu@google.com>2014-07-15 13:58:38 -0700
committerBo Liu <boliu@google.com>2014-07-15 13:58:38 -0700
commitd8768460803a85d8661d90a648107e03e642b108 (patch)
treedac74b0e13038ffe7b23068354083630442bd69e
parent8e557e77c7498aaa0d971db28b131b5fa4114c12 (diff)
parent1b4ed025b291d5d226ee45f1f1211becd39a3560 (diff)
downloadwebrtc-d8768460803a85d8661d90a648107e03e642b108.tar.gz
Merge third_party/webrtc from https://chromium.googlesource.com/a/external/webrtc/stable/webrtc.git at 1b4ed025b291d5d226ee45f1f1211becd39a3560
This commit was generated by merge_from_chromium.py. Change-Id: Id66890959e976958d482471269ed19c042e056d1
-rw-r--r--modules/rtp_rtcp/source/rtp_sender.h2
1 files changed, 1 insertions, 1 deletions
diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index 17b026ca..291e619b 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -397,7 +397,7 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer {
// that by the time the function returns there is no guarantee
// that the target bitrate is still valid.
scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
- uint16_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
+ uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
};
} // namespace webrtc