Lint fix for webrtc/modules/video_coding PART 1!

webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.cc

 /*
  *  Copyright (c) 2015 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 136 matching lines @@
     header->fragmentationOffset[i] = frag_offsets[i];
     header->fragmentationLength[i] = frag_lengths[i];
     header->fragmentationPlType[i] = 0;
     header->fragmentationTimeDiff[i] = 0;
   }
   *out_header = header.release();
   CFRelease(contiguous_buffer);
   return true;
 }
 
-bool H264AnnexBBufferToCMSampleBuffer(
-    const uint8_t* annexb_buffer,
-    size_t annexb_buffer_size,
-    CMVideoFormatDescriptionRef video_format,
-    CMSampleBufferRef* out_sample_buffer) {
+bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
+                                      size_t annexb_buffer_size,
+                                      CMVideoFormatDescriptionRef video_format,
+                                      CMSampleBufferRef* out_sample_buffer) {
   RTC_DCHECK(annexb_buffer);
   RTC_DCHECK(out_sample_buffer);
   *out_sample_buffer = nullptr;
 
   // The buffer we receive via RTP has 00 00 00 01 start code artifically
   // embedded by the RTP depacketizer. Extract NALU information.
   // TODO(tkchin): handle potential case where sps and pps are delivered
   // separately.
   uint8_t first_nalu_type = annexb_buffer[4] & 0x1f;
   bool is_first_nalu_type_sps = first_nalu_type == 0x7;
@@ skipping 175 matching lines @@
   return true;
 }
 
 size_t AvccBufferWriter::BytesRemaining() const {
   return length_ - offset_;
 }
 
 }  // namespace webrtc
 
 #endif  // defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)