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Unified Diff: webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc

Issue 1422023002: Use webrtc/base/logging.h for rtp_rtcp. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 2 months ago
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Index: webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc
diff --git a/webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc b/webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc
index 00ab9d4a6593d391f8efce5e8520662bc4d5a0f7..2fb723e3f916e45704cdecb06a23c62b71dceefe 100644
--- a/webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc
+++ b/webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc
@@ -10,9 +10,9 @@
#include "webrtc/modules/rtp_rtcp/source/h264_sps_parser.h"
-#include "webrtc/base/bytebuffer.h"
#include "webrtc/base/bitbuffer.h"
-#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/base/bytebuffer.h"
+#include "webrtc/base/logging.h"
#define RETURN_FALSE_ON_FAIL(x) \
if (!(x)) { \

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