Rename constants files in webrtc/{media,p2p}

webrtc/pc/mediasession_unittest.cc

 /*
  *  Copyright 2004 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.
  */
 
 #include <string>
 #include <vector>
 
 #include "webrtc/base/fakesslidentity.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/messagedigest.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/media/base/codec.h"
 #include "webrtc/media/base/testutils.h"
-#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/p2pconstants.h"
 #include "webrtc/p2p/base/transportdescription.h"
 #include "webrtc/p2p/base/transportinfo.h"
 #include "webrtc/pc/mediasession.h"
 #include "webrtc/pc/srtpfilter.h"
 
 #ifdef HAVE_SRTP
 #define ASSERT_CRYPTO(cd, s, cs) \
     ASSERT_EQ(cricket::CT_NONE, cd->crypto_required()); \
     ASSERT_EQ(s, cd->cryptos().size()); \
     ASSERT_EQ(std::string(cs), cd->cryptos()[0].cipher_suite)
@@ skipping 2287 matching lines @@
   const ContentInfo* audio_content = GetFirstAudioContent(updated_offer.get());
   const ContentInfo* video_content = GetFirstVideoContent(updated_offer.get());
   const ContentInfo* data_content = GetFirstDataContent(updated_offer.get());
   ASSERT_TRUE(audio_content != nullptr);
   ASSERT_TRUE(video_content != nullptr);
   ASSERT_TRUE(data_content != nullptr);
   EXPECT_EQ("audio_modified", audio_content->name);
   EXPECT_EQ("video_modified", video_content->name);
   EXPECT_EQ("data_modified", data_content->name);
 }