webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc - Issue 1422023002: Use webrtc/base/logging.h for rtp_rtcp.

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

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc

index 47a436309a3f144b15664878dc8a894a716e6eca..619833223d05c01cebfbdb0c87eb4a34f9d54e86 100644

--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc

+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc

@@ -14,9 +14,9 @@

#include <math.h> // pow()

#include <string.h> // memcpy()

+#include "webrtc/base/logging.h"

#include "webrtc/base/trace_event.h"

#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"

-#include "webrtc/system_wrappers/interface/logging.h"

namespace webrtc {

RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(