Move talk/media to webrtc/media

I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.

The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.

The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.

License headers will be updated in a follow-up CL in order to not
break Git history.

Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
  except for these files:
  talk/app/webrtc/peerconnectionendtoend_unittest.cc
  talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
  talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
  webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
  webrtc/base/testutils.cc
  webrtc/base/testutils.h

The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.

I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.

For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/

BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1587193006

Cr-Commit-Position: refs/heads/master@{#11495}
diff --git a/webrtc/api/api.gyp b/webrtc/api/api.gyp
index ec51b47..ac4ea84 100644
--- a/webrtc/api/api.gyp
+++ b/webrtc/api/api.gyp
@@ -91,7 +91,17 @@
                     '-framework QuartzCore',
                   ]
                 }
-              }
+              },
+              # TODO(kjellander): Make the code compile without disabling these.
+              # See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
+              'cflags': [
+                '-Wno-return-type',
+              ],
+              'xcode_settings': {
+                'WARNING_CFLAGS': [
+                  '-Wno-return-type',
+                ],
+              },
             }],
             ['OS=="mac"', {
               'sources': [
diff --git a/webrtc/api/objc/RTCVideoFrame+Private.h b/webrtc/api/objc/RTCVideoFrame+Private.h
index 954344a..873d3eb 100644
--- a/webrtc/api/objc/RTCVideoFrame+Private.h
+++ b/webrtc/api/objc/RTCVideoFrame+Private.h
@@ -10,7 +10,7 @@
 
 #import "RTCVideoFrame.h"
 
-#include "talk/media/base/videoframe.h"
+#include "webrtc/media/base/videoframe.h"
 
 NS_ASSUME_NONNULL_BEGIN
 
diff --git a/webrtc/api/objc/avfoundationvideocapturer.h b/webrtc/api/objc/avfoundationvideocapturer.h
index 79dec58..2ee456d 100644
--- a/webrtc/api/objc/avfoundationvideocapturer.h
+++ b/webrtc/api/objc/avfoundationvideocapturer.h
@@ -11,8 +11,8 @@
 #ifndef WEBRTC_API_OBJC_AVFOUNDATION_VIDEO_CAPTURER_H_
 #define WEBRTC_API_OBJC_AVFOUNDATION_VIDEO_CAPTURER_H_
 
-#include "talk/media/base/videocapturer.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/media/base/videocapturer.h"
 #include "webrtc/video_frame.h"
 
 #import <AVFoundation/AVFoundation.h>