Remove ViE lint warnings that should have been caught at upload time.
TEST=cpplint.py video_engine/*
Review URL: https://webrtc-codereview.appspot.com/964018
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3151 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/video_engine/vie_encoder.cc b/video_engine/vie_encoder.cc
index df8ba91..36ad862 100644
--- a/video_engine/vie_encoder.cc
+++ b/video_engine/vie_encoder.cc
@@ -67,7 +67,7 @@
}
virtual void TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number,
int64_t capture_time_ms) {
- owner_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms);
+ owner_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms);
}
virtual void TimeToSendPadding(int /*bytes*/) {
// TODO(pwestin): Hook up this.
diff --git a/video_engine/vie_remb.h b/video_engine/vie_remb.h
index 63e9638..0ad36ca 100644
--- a/video_engine/vie_remb.h
+++ b/video_engine/vie_remb.h
@@ -8,19 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-// 1. Register a RtpRtcp module to include in the REMB packet.
-// 2. When UpdateBitrateEstimate is called for the first time for a SSRC, add it
-// to the map.
-// 3. Send a new REMB every kRembSendIntervallMs or if a lower bitrate estimate
-// for a specified SSRC.
-
-
#ifndef WEBRTC_VIDEO_ENGINE_VIE_REMB_H_
#define WEBRTC_VIDEO_ENGINE_VIE_REMB_H_
#include <list>
-#include <map>
#include <utility>
+#include <vector>
#include "modules/interface/module.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
diff --git a/video_engine/vie_remb_unittest.cc b/video_engine/vie_remb_unittest.cc
index 680a892..4cd1cc1 100644
--- a/video_engine/vie_remb_unittest.cc
+++ b/video_engine/vie_remb_unittest.cc
@@ -14,6 +14,8 @@
#include <gmock/gmock.h>
#include <gtest/gtest.h>
+#include <vector>
+
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "webrtc/modules/utility/interface/process_thread.h"