Lint clean video/ and add lint presubmit check.

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 382 matching lines @@
     RTC_DCHECK_GT(streams[i].width, 0u);
     RTC_DCHECK_GT(streams[i].height, 0u);
     RTC_DCHECK_GT(streams[i].max_framerate, 0);
     // Different framerates not supported per stream at the moment.
     RTC_DCHECK_EQ(streams[i].max_framerate, streams[0].max_framerate);
     RTC_DCHECK_GE(streams[i].min_bitrate_bps, 0);
     RTC_DCHECK_GE(streams[i].target_bitrate_bps, streams[i].min_bitrate_bps);
     RTC_DCHECK_GE(streams[i].max_bitrate_bps, streams[i].target_bitrate_bps);
     RTC_DCHECK_GE(streams[i].max_qp, 0);
 
-    sim_stream->width = static_cast<unsigned short>(streams[i].width);
-    sim_stream->height = static_cast<unsigned short>(streams[i].height);
+    sim_stream->width = static_cast<uint16_t>(streams[i].width);
+    sim_stream->height = static_cast<uint16_t>(streams[i].height);
     sim_stream->minBitrate = streams[i].min_bitrate_bps / 1000;
     sim_stream->targetBitrate = streams[i].target_bitrate_bps / 1000;
     sim_stream->maxBitrate = streams[i].max_bitrate_bps / 1000;
     sim_stream->qpMax = streams[i].max_qp;
     sim_stream->numberOfTemporalLayers = static_cast<unsigned char>(
         streams[i].temporal_layer_thresholds_bps.size() + 1);
 
     video_codec.width = std::max(video_codec.width,
-                                 static_cast<unsigned short>(streams[i].width));
+                                 static_cast<uint16_t>(streams[i].width));
     video_codec.height = std::max(
-        video_codec.height, static_cast<unsigned short>(streams[i].height));
+        video_codec.height, static_cast<uint16_t>(streams[i].height));
     video_codec.minBitrate =
-        std::min(video_codec.minBitrate,
-                 static_cast<unsigned int>(streams[i].min_bitrate_bps / 1000));
+        std::min(static_cast<uint16_t>(video_codec.minBitrate),
+                 static_cast<uint16_t>(streams[i].min_bitrate_bps / 1000));
     video_codec.maxBitrate += streams[i].max_bitrate_bps / 1000;
     video_codec.qpMax = std::max(video_codec.qpMax,
                                  static_cast<unsigned int>(streams[i].max_qp));
   }
 
   // Set to zero to not update the bitrate controller from ViEEncoder, as
   // the bitrate controller is already set from Call.
   video_codec.startBitrate = 0;
 
   RTC_DCHECK_GT(streams[0].max_framerate, 0);
@@ skipping 64 matching lines @@
 
   RTC_DCHECK_GE(config_.rtp.rtx.payload_type, 0);
   vie_channel_->SetRtxSendPayloadType(config_.rtp.rtx.payload_type,
                                       config_.encoder_settings.payload_type);
 }
 
 std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const {
   std::map<uint32_t, RtpState> rtp_states;
   for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.ssrcs[i];
-    rtp_states[ssrc] = vie_channel_->GetRtpStateForSsrc( ssrc);
+    rtp_states[ssrc] = vie_channel_->GetRtpStateForSsrc(ssrc);
   }
 
   for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
     rtp_states[ssrc] = vie_channel_->GetRtpStateForSsrc(ssrc);
   }
 
   return rtp_states;
 }
 
@@ skipping 61 matching lines @@
   vie_encoder_->SetSsrcs(used_ssrcs);
 
   // Restart the media flow
   vie_encoder_->Restart();
 
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc