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}

.gitignore

@@ -50,13 +50,16 @@
 /out
 /resources/**/*.aecdump
 /resources/**/*.bin
+/resources/**/*.byte
 /resources/**/*.chn
 /resources/**/*.dat
+/resources/**/*.frames
 /resources/**/*.gai
 /resources/**/*.jpg
 /resources/**/*.pcap
 /resources/**/*.pcm
 /resources/**/*.rtp
+/resources/**/*.rtpdump
 /resources/**/*.rx
 /resources/**/*.wav
 /resources/**/*.yuv

DEPS

@@ -42,6 +42,7 @@ include_rules = [
   '+testing',
   '+third_party',
   '+unicode',
+  '+usrsctplib',
   '+webrtc',
   '+vpx',
 ]

resources/media/1.frame_plus_1.byte.sha1

@@ -0,0 +1 @@
+8c398a382d0a36e733969230be593682c9a0dd75

resources/media/captured-320x240-2s-48.frames.sha1

@@ -0,0 +1 @@
+bd6052a1dff9c55255034cbac5f98c0cb17be20b

resources/media/faces.1280x720_P420.yuv.sha1

@@ -0,0 +1 @@
+5ffdf950177ecd872e1f7f21c8d26936b950e09a

resources/media/faces_I400.jpg.sha1

@@ -0,0 +1 @@
+3c7b95f86c7c219d959c04bbdf1db35b7faf8847

resources/media/faces_I411.jpg.sha1

@@ -0,0 +1 @@
+4ab8170541a21fb7015496e84837a8f801077cfd

resources/media/faces_I420.jpg.sha1

@@ -0,0 +1 @@
+6e074b266e6c6ef0c3290aa3101487009f42b245

resources/media/faces_I422.jpg.sha1

@@ -0,0 +1 @@
+4d25abcbe6d2d71a29e52b5eab8a127a1f605359

resources/media/faces_I444.jpg.sha1

@@ -0,0 +1 @@
+4e040d5332058d3435091ef441af7385198c2029

resources/media/h264-svc-99-640x360.rtpdump.sha1

@@ -0,0 +1 @@
+41ae53fcad7343b66c8351097a071bbb25e47ba4

resources/media/video.rtpdump.sha1

@@ -0,0 +1 @@
+714fa3ce8101fa7baa61842e3d8b37ebb2d602aa

resources/media/voice.rtpdump.sha1

@@ -0,0 +1 @@
+bf76337737b1ed6d6815f6f3eeeee1e25dce55b6

talk/app/webrtc/androidvideocapturer.cc

@@ -27,10 +27,10 @@
 #include "talk/app/webrtc/androidvideocapturer.h"
 
 #include "talk/app/webrtc/java/jni/native_handle_impl.h"
-#include "talk/media/webrtc/webrtcvideoframe.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/json.h"
 #include "webrtc/base/timeutils.h"
+#include "webrtc/media/webrtc/webrtcvideoframe.h"
 
 namespace webrtc {
 

talk/app/webrtc/androidvideocapturer.h

@@ -30,9 +30,9 @@
 #include <string>
 #include <vector>
 
-#include "talk/media/base/videocapturer.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/common_video/include/video_frame_buffer.h"
+#include "webrtc/media/base/videocapturer.h"
 
 namespace webrtc {
 

talk/app/webrtc/datachannel.cc

@@ -31,9 +31,9 @@
 
 #include "talk/app/webrtc/mediastreamprovider.h"
 #include "talk/app/webrtc/sctputils.h"
-#include "talk/media/sctp/sctpdataengine.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/refcount.h"
+#include "webrtc/media/sctp/sctpdataengine.h"
 
 namespace webrtc {
 

talk/app/webrtc/datachannel.h

@@ -34,11 +34,11 @@
 
 #include "talk/app/webrtc/datachannelinterface.h"
 #include "talk/app/webrtc/proxy.h"
-#include "talk/media/base/mediachannel.h"
 #include "talk/session/media/channel.h"
 #include "webrtc/base/messagehandler.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/sigslot.h"
+#include "webrtc/media/base/mediachannel.h"
 
 namespace webrtc {
 

talk/app/webrtc/java/jni/androidmediacodeccommon.h

@@ -36,6 +36,7 @@
 #include "talk/app/webrtc/java/jni/jni_helpers.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/thread.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc_jni {

talk/app/webrtc/java/jni/androidmediadecoder_jni.cc

@@ -30,10 +30,15 @@
 #include <vector>
 
 #include "talk/app/webrtc/java/jni/androidmediadecoder_jni.h"
+// NOTICE: androidmediadecoder_jni.h must be included before
+// androidmediacodeccommon.h to avoid build errors.
 #include "talk/app/webrtc/java/jni/androidmediacodeccommon.h"
 #include "talk/app/webrtc/java/jni/classreferenceholder.h"
 #include "talk/app/webrtc/java/jni/native_handle_impl.h"
 #include "talk/app/webrtc/java/jni/surfacetexturehelper_jni.h"
+#include "third_party/libyuv/include/libyuv/convert.h"
+#include "third_party/libyuv/include/libyuv/convert_from.h"
+#include "third_party/libyuv/include/libyuv/video_common.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
@@ -44,9 +49,6 @@
 #include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/system_wrappers/include/logcat_trace_context.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
-#include "third_party/libyuv/include/libyuv/convert.h"
-#include "third_party/libyuv/include/libyuv/convert_from.h"
-#include "third_party/libyuv/include/libyuv/video_common.h"
 
 using rtc::Bind;
 using rtc::Thread;

talk/app/webrtc/java/jni/androidmediadecoder_jni.h

@@ -30,7 +30,7 @@
 #define TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIADECODER_JNI_H_
 
 #include "talk/app/webrtc/java/jni/eglbase_jni.h"
-#include "talk/media/webrtc/webrtcvideodecoderfactory.h"
+#include "webrtc/media/webrtc/webrtcvideodecoderfactory.h"
 
 namespace webrtc_jni {
 

talk/app/webrtc/java/jni/androidmediaencoder_jni.cc

@@ -27,9 +27,14 @@
  */
 
 #include "talk/app/webrtc/java/jni/androidmediaencoder_jni.h"
-#include "talk/app/webrtc/java/jni/classreferenceholder.h"
+// NOTICE: androidmediaencoder_jni.h must be included before
+// androidmediacodeccommon.h to avoid build errors.
 #include "talk/app/webrtc/java/jni/androidmediacodeccommon.h"
+#include "talk/app/webrtc/java/jni/classreferenceholder.h"
 #include "talk/app/webrtc/java/jni/native_handle_impl.h"
+#include "third_party/libyuv/include/libyuv/convert.h"
+#include "third_party/libyuv/include/libyuv/convert_from.h"
+#include "third_party/libyuv/include/libyuv/video_common.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
@@ -42,9 +47,6 @@
 #include "webrtc/modules/video_coding/utility/vp8_header_parser.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/system_wrappers/include/logcat_trace_context.h"
-#include "third_party/libyuv/include/libyuv/convert.h"
-#include "third_party/libyuv/include/libyuv/convert_from.h"
-#include "third_party/libyuv/include/libyuv/video_common.h"
 
 using rtc::Bind;
 using rtc::Thread;

talk/app/webrtc/java/jni/androidmediaencoder_jni.h

@@ -32,7 +32,7 @@
 #include <vector>
 
 #include "talk/app/webrtc/java/jni/eglbase_jni.h"
-#include "talk/media/webrtc/webrtcvideoencoderfactory.h"
+#include "webrtc/media/webrtc/webrtcvideoencoderfactory.h"
 
 namespace webrtc_jni {
 

talk/app/webrtc/java/jni/androidnetworkmonitor_jni.cc

@@ -29,11 +29,11 @@
 
 #include <dlfcn.h>
 
+#include "talk/app/webrtc/java/jni/classreferenceholder.h"
+#include "talk/app/webrtc/java/jni/jni_helpers.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/ipaddress.h"
-#include "talk/app/webrtc/java/jni/classreferenceholder.h"
-#include "talk/app/webrtc/java/jni/jni_helpers.h"
 
 namespace webrtc_jni {
 

talk/app/webrtc/java/jni/androidnetworkmonitor_jni.h

@@ -32,9 +32,9 @@
 
 #include <map>
 
+#include "talk/app/webrtc/java/jni/jni_helpers.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/thread_checker.h"
-#include "talk/app/webrtc/java/jni/jni_helpers.h"
 
 namespace webrtc_jni {
 

talk/app/webrtc/java/jni/classreferenceholder.h

@@ -35,8 +35,8 @@
 #define TALK_APP_WEBRTC_JAVA_JNI_CLASSREFERENCEHOLDER_H_
 
 #include <jni.h>
-#include <string>
 #include <map>
+#include <string>
 
 namespace webrtc_jni {
 

talk/app/webrtc/java/jni/native_handle_impl.cc

@@ -31,9 +31,9 @@
 #include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/keep_ref_until_done.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/base/logging.h"
 
 using webrtc::NativeHandleBuffer;
 

talk/app/webrtc/java/jni/peerconnection_jni.cc

@@ -59,20 +59,15 @@
 #include <limits>
 #include <utility>
 
+#include "talk/app/webrtc/dtlsidentitystore.h"
 #include "talk/app/webrtc/java/jni/classreferenceholder.h"
 #include "talk/app/webrtc/java/jni/jni_helpers.h"
 #include "talk/app/webrtc/java/jni/native_handle_impl.h"
-#include "talk/app/webrtc/dtlsidentitystore.h"
 #include "talk/app/webrtc/mediaconstraintsinterface.h"
 #include "talk/app/webrtc/peerconnectioninterface.h"
 #include "talk/app/webrtc/rtpreceiverinterface.h"
 #include "talk/app/webrtc/rtpsenderinterface.h"
 #include "talk/app/webrtc/videosourceinterface.h"
-#include "talk/media/base/videocapturer.h"
-#include "talk/media/base/videorenderer.h"
-#include "talk/media/devices/videorendererfactory.h"
-#include "talk/media/webrtc/webrtcvideodecoderfactory.h"
-#include "talk/media/webrtc/webrtcvideoencoderfactory.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event_tracer.h"
@@ -82,6 +77,11 @@
 #include "webrtc/base/networkmonitor.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/stringutils.h"
+#include "webrtc/media/base/videocapturer.h"
+#include "webrtc/media/base/videorenderer.h"
+#include "webrtc/media/devices/videorendererfactory.h"
+#include "webrtc/media/webrtc/webrtcvideodecoderfactory.h"
+#include "webrtc/media/webrtc/webrtcvideoencoderfactory.h"
 #include "webrtc/system_wrappers/include/field_trial_default.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_base.h"
@@ -90,8 +90,8 @@
 #include "talk/app/webrtc/androidvideocapturer.h"
 #include "talk/app/webrtc/java/jni/androidmediadecoder_jni.h"
 #include "talk/app/webrtc/java/jni/androidmediaencoder_jni.h"
-#include "talk/app/webrtc/java/jni/androidvideocapturer_jni.h"
 #include "talk/app/webrtc/java/jni/androidnetworkmonitor_jni.h"
+#include "talk/app/webrtc/java/jni/androidvideocapturer_jni.h"
 #include "webrtc/modules/video_render/video_render_internal.h"
 #include "webrtc/system_wrappers/include/logcat_trace_context.h"
 using webrtc::LogcatTraceContext;

talk/app/webrtc/jsepicecandidate.h

@@ -33,8 +33,8 @@
 #include <string>
 
 #include "talk/app/webrtc/jsep.h"
-#include "webrtc/p2p/base/candidate.h"
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/p2p/base/candidate.h"
 
 namespace webrtc {
 

talk/app/webrtc/jsepsessiondescription_unittest.cc

@@ -29,15 +29,15 @@
 
 #include "talk/app/webrtc/jsepicecandidate.h"
 #include "talk/app/webrtc/jsepsessiondescription.h"
-#include "webrtc/p2p/base/candidate.h"
-#include "webrtc/p2p/base/constants.h"
-#include "webrtc/p2p/base/sessiondescription.h"
 #include "talk/session/media/mediasession.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/stringencode.h"
+#include "webrtc/p2p/base/candidate.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/sessiondescription.h"
 
 using webrtc::IceCandidateCollection;
 using webrtc::IceCandidateInterface;

talk/app/webrtc/localaudiosource.cc

@@ -30,7 +30,7 @@
 #include <vector>
 
 #include "talk/app/webrtc/mediaconstraintsinterface.h"
-#include "talk/media/base/mediaengine.h"
+#include "webrtc/media/base/mediaengine.h"
 
 using webrtc::MediaConstraintsInterface;
 using webrtc::MediaSourceInterface;

talk/app/webrtc/localaudiosource.h

@@ -31,8 +31,8 @@
 #include "talk/app/webrtc/mediastreaminterface.h"
 #include "talk/app/webrtc/notifier.h"
 #include "talk/app/webrtc/peerconnectioninterface.h"
-#include "talk/media/base/mediachannel.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/media/base/mediachannel.h"
 
 // LocalAudioSource implements AudioSourceInterface.
 // This contains settings for switching audio processing on and off.

talk/app/webrtc/localaudiosource_unittest.cc

@@ -31,9 +31,9 @@
 #include <vector>
 
 #include "talk/app/webrtc/test/fakeconstraints.h"
-#include "talk/media/base/fakemediaengine.h"
-#include "talk/media/base/fakevideorenderer.h"
 #include "webrtc/base/gunit.h"
+#include "webrtc/media/base/fakemediaengine.h"
+#include "webrtc/media/base/fakevideorenderer.h"
 
 using webrtc::LocalAudioSource;
 using webrtc::MediaConstraintsInterface;

talk/app/webrtc/objc/RTCI420Frame+Internal.h

@@ -27,7 +27,7 @@
 
 #import "RTCI420Frame.h"
 
-#include "talk/media/base/videoframe.h"
+#include "webrtc/media/base/videoframe.h"
 
 @interface RTCI420Frame (Internal)
 

talk/app/webrtc/objc/RTCI420Frame.mm

@@ -27,7 +27,7 @@
 
 #import "RTCI420Frame.h"
 
-#include "talk/media/base/videoframe.h"
+#include "webrtc/media/base/videoframe.h"
 #include "webrtc/base/scoped_ptr.h"
 
 @implementation RTCI420Frame {

talk/app/webrtc/objc/RTCVideoCapturer.mm

@@ -31,8 +31,8 @@
 
 #import "RTCVideoCapturer+Internal.h"
 
-#include "talk/media/base/videocapturer.h"
+#include "webrtc/media/base/videocapturer.h"
-#include "talk/media/devices/devicemanager.h"
+#include "webrtc/media/devices/devicemanager.h"
 
 @implementation RTCVideoCapturer {
   rtc::scoped_ptr<cricket::VideoCapturer> _capturer;

talk/app/webrtc/objc/avfoundationvideocapturer.h

@@ -28,8 +28,8 @@
 #ifndef TALK_APP_WEBRTC_OBJC_AVFOUNDATION_VIDEO_CAPTURER_H_
 #define TALK_APP_WEBRTC_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>

talk/app/webrtc/peerconnection.cc

@@ -48,13 +48,13 @@
 #include "talk/app/webrtc/streamcollection.h"
 #include "talk/app/webrtc/videosource.h"
 #include "talk/app/webrtc/videotrack.h"
-#include "talk/media/sctp/sctpdataengine.h"
 #include "talk/session/media/channelmanager.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/trace_event.h"
+#include "webrtc/media/sctp/sctpdataengine.h"
 #include "webrtc/p2p/client/basicportallocator.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 

talk/app/webrtc/peerconnection_unittest.cc

@@ -47,7 +47,6 @@
 #include "talk/app/webrtc/test/fakevideotrackrenderer.h"
 #include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
 #include "talk/app/webrtc/videosourceinterface.h"
-#include "talk/media/webrtc/fakewebrtcvideoengine.h"
 #include "talk/session/media/mediasession.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/physicalsocketserver.h"
@@ -56,6 +55,7 @@
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/virtualsocketserver.h"
+#include "webrtc/media/webrtc/fakewebrtcvideoengine.h"
 #include "webrtc/p2p/base/constants.h"
 #include "webrtc/p2p/base/sessiondescription.h"
 #include "webrtc/p2p/client/fakeportallocator.h"

talk/app/webrtc/peerconnectionendtoend_unittest.cc

@@ -26,6 +26,7 @@
  */
 
 #include "talk/app/webrtc/test/peerconnectiontestwrapper.h"
+// Notice that mockpeerconnectionobservers.h must be included after the above!
 #include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
 #ifdef WEBRTC_ANDROID
 #include "talk/app/webrtc/test/androidtestinitializer.h"

talk/app/webrtc/peerconnectionfactory.cc

@@ -40,10 +40,10 @@
 #include "talk/app/webrtc/videosource.h"
 #include "talk/app/webrtc/videosourceproxy.h"
 #include "talk/app/webrtc/videotrack.h"
-#include "talk/media/webrtc/webrtcmediaengine.h"
-#include "talk/media/webrtc/webrtcvideodecoderfactory.h"
-#include "talk/media/webrtc/webrtcvideoencoderfactory.h"
 #include "webrtc/base/bind.h"
+#include "webrtc/media/webrtc/webrtcmediaengine.h"
+#include "webrtc/media/webrtc/webrtcvideodecoderfactory.h"
+#include "webrtc/media/webrtc/webrtcvideoencoderfactory.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/p2p/base/basicpacketsocketfactory.h"
 #include "webrtc/p2p/client/basicportallocator.h"

talk/app/webrtc/peerconnectionfactory_unittest.cc

@@ -36,12 +36,12 @@
 #include "talk/app/webrtc/test/fakedtlsidentitystore.h"
 #include "talk/app/webrtc/test/fakevideotrackrenderer.h"
 #include "talk/app/webrtc/videosourceinterface.h"
-#include "talk/media/base/fakevideocapturer.h"
-#include "talk/media/webrtc/webrtccommon.h"
-#include "talk/media/webrtc/webrtcvoe.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread.h"
+#include "webrtc/media/base/fakevideocapturer.h"
+#include "webrtc/media/webrtc/webrtccommon.h"
+#include "webrtc/media/webrtc/webrtcvoe.h"
 #include "webrtc/p2p/client/fakeportallocator.h"
 
 using webrtc::DataChannelInterface;

talk/app/webrtc/peerconnectioninterface.h

@@ -74,8 +74,8 @@
 
 #include "talk/app/webrtc/datachannelinterface.h"
 #include "talk/app/webrtc/dtlsidentitystore.h"
-#include "talk/app/webrtc/dtmfsenderinterface.h"
 #include "talk/app/webrtc/dtlsidentitystore.h"
+#include "talk/app/webrtc/dtmfsenderinterface.h"
 #include "talk/app/webrtc/jsep.h"
 #include "talk/app/webrtc/mediastreaminterface.h"
 #include "talk/app/webrtc/rtpreceiverinterface.h"
@@ -85,8 +85,8 @@
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/network.h"
 #include "webrtc/base/rtccertificate.h"
-#include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/socketaddress.h"
+#include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/p2p/base/portallocator.h"
 
 namespace rtc {

talk/app/webrtc/peerconnectioninterface_unittest.cc

@@ -46,8 +46,6 @@
 #include "talk/app/webrtc/test/testsdpstrings.h"
 #include "talk/app/webrtc/videosource.h"
 #include "talk/app/webrtc/videotrack.h"
-#include "talk/media/base/fakevideocapturer.h"
-#include "talk/media/sctp/sctpdataengine.h"
 #include "talk/session/media/mediasession.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/scoped_ptr.h"
@@ -55,6 +53,8 @@
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/thread.h"
+#include "webrtc/media/base/fakevideocapturer.h"
+#include "webrtc/media/sctp/sctpdataengine.h"
 #include "webrtc/p2p/client/fakeportallocator.h"
 
 static const char kStreamLabel1[] = "local_stream_1";

talk/app/webrtc/remoteaudiosource.h

@@ -33,9 +33,9 @@
 
 #include "talk/app/webrtc/mediastreaminterface.h"
 #include "talk/app/webrtc/notifier.h"
-#include "talk/media/base/audiorenderer.h"
 #include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/media/base/audiorenderer.h"
 
 namespace rtc {
 struct Message;

talk/app/webrtc/remotevideocapturer.cc

@@ -27,8 +27,8 @@
 
 #include "talk/app/webrtc/remotevideocapturer.h"
 
-#include "talk/media/base/videoframe.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/media/base/videoframe.h"
 
 namespace webrtc {
 

talk/app/webrtc/remotevideocapturer.h

@@ -31,8 +31,8 @@
 #include <vector>
 
 #include "talk/app/webrtc/mediastreaminterface.h"
-#include "talk/media/base/videocapturer.h"
+#include "webrtc/media/base/videocapturer.h"
-#include "talk/media/base/videorenderer.h"
+#include "webrtc/media/base/videorenderer.h"
 
 namespace webrtc {
 

talk/app/webrtc/remotevideocapturer_unittest.cc

@@ -28,8 +28,8 @@
 #include <string>
 
 #include "talk/app/webrtc/remotevideocapturer.h"
-#include "talk/media/webrtc/webrtcvideoframe.h"
 #include "webrtc/base/gunit.h"
+#include "webrtc/media/webrtc/webrtcvideoframe.h"
 
 using cricket::CaptureState;
 using cricket::VideoCapturer;

talk/app/webrtc/rtpreceiverinterface.h

@@ -33,8 +33,8 @@
 
 #include <string>
 
-#include "talk/app/webrtc/proxy.h"
 #include "talk/app/webrtc/mediastreaminterface.h"
+#include "talk/app/webrtc/proxy.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 

talk/app/webrtc/rtpsender.h

@@ -37,10 +37,10 @@
 #include "talk/app/webrtc/mediastreamprovider.h"
 #include "talk/app/webrtc/rtpsenderinterface.h"
 #include "talk/app/webrtc/statscollector.h"
-#include "talk/media/base/audiorenderer.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/media/base/audiorenderer.h"
 
 namespace webrtc {
 

talk/app/webrtc/rtpsenderinterface.h

@@ -33,8 +33,8 @@
 
 #include <string>
 
-#include "talk/app/webrtc/proxy.h"
 #include "talk/app/webrtc/mediastreaminterface.h"
+#include "talk/app/webrtc/proxy.h"
 #include "talk/session/media/mediasession.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"

talk/app/webrtc/rtpsenderreceiver_unittest.cc

@@ -36,11 +36,11 @@
 #include "talk/app/webrtc/streamcollection.h"
 #include "talk/app/webrtc/videosource.h"
 #include "talk/app/webrtc/videotrack.h"
-#include "talk/media/base/fakevideocapturer.h"
-#include "talk/media/base/mediachannel.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/gunit.h"
+#include "webrtc/media/base/fakevideocapturer.h"
+#include "webrtc/media/base/mediachannel.h"
 
 using ::testing::_;
 using ::testing::Exactly;

talk/app/webrtc/statscollector_unittest.cc

@@ -31,14 +31,13 @@
 
 #include "talk/app/webrtc/statscollector.h"
 
-#include "talk/app/webrtc/peerconnection.h"
-#include "talk/app/webrtc/peerconnectionfactory.h"
 #include "talk/app/webrtc/mediastream.h"
 #include "talk/app/webrtc/mediastreaminterface.h"
 #include "talk/app/webrtc/mediastreamtrack.h"
+#include "talk/app/webrtc/peerconnection.h"
+#include "talk/app/webrtc/peerconnectionfactory.h"
 #include "talk/app/webrtc/test/fakedatachannelprovider.h"
 #include "talk/app/webrtc/videotrack.h"
-#include "talk/media/base/fakemediaengine.h"
 #include "talk/session/media/channelmanager.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
@@ -46,6 +45,7 @@
 #include "webrtc/base/fakesslidentity.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/network.h"
+#include "webrtc/media/base/fakemediaengine.h"
 #include "webrtc/p2p/base/faketransportcontroller.h"
 
 using rtc::scoped_ptr;

talk/app/webrtc/statstypes.h

@@ -38,9 +38,9 @@
 
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/common.h"
+#include "webrtc/base/linked_ptr.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/linked_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/thread_checker.h"

talk/app/webrtc/test/fakeperiodicvideocapturer.h

@@ -31,8 +31,8 @@
 #ifndef TALK_APP_WEBRTC_TEST_FAKEPERIODICVIDEOCAPTURER_H_
 #define TALK_APP_WEBRTC_TEST_FAKEPERIODICVIDEOCAPTURER_H_
 
-#include "talk/media/base/fakevideocapturer.h"
 #include "webrtc/base/thread.h"
+#include "webrtc/media/base/fakevideocapturer.h"
 
 namespace webrtc {
 

talk/app/webrtc/test/fakevideotrackrenderer.h

@@ -29,7 +29,7 @@
 #define TALK_APP_WEBRTC_TEST_FAKEVIDEOTRACKRENDERER_H_
 
 #include "talk/app/webrtc/mediastreaminterface.h"
-#include "talk/media/base/fakevideorenderer.h"
+#include "webrtc/media/base/fakevideorenderer.h"
 
 namespace webrtc {
 

talk/app/webrtc/videosource.cc

@@ -27,8 +27,8 @@
 
 #include "talk/app/webrtc/videosource.h"
 
-#include <vector>
 #include <cstdlib>
+#include <vector>
 
 #include "talk/app/webrtc/mediaconstraintsinterface.h"
 #include "talk/session/media/channelmanager.h"

talk/app/webrtc/videosource.h

@@ -34,11 +34,11 @@
 #include "talk/app/webrtc/notifier.h"
 #include "talk/app/webrtc/videosourceinterface.h"
 #include "talk/app/webrtc/videotrackrenderers.h"
-#include "talk/media/base/videocapturer.h"
-#include "talk/media/base/videocommon.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/media/base/videosinkinterface.h"
+#include "webrtc/media/base/videocapturer.h"
+#include "webrtc/media/base/videocommon.h"
 
 // VideoSource implements VideoSourceInterface. It owns a
 // cricket::VideoCapturer and make sure the camera is started at a resolution

talk/app/webrtc/videosource_unittest.cc

@@ -31,12 +31,12 @@
 #include "talk/app/webrtc/remotevideocapturer.h"
 #include "talk/app/webrtc/test/fakeconstraints.h"
 #include "talk/app/webrtc/videosource.h"
-#include "talk/media/base/fakemediaengine.h"
-#include "talk/media/base/fakevideocapturer.h"
-#include "talk/media/base/fakevideorenderer.h"
-#include "talk/media/webrtc/webrtcvideoframe.h"
 #include "talk/session/media/channelmanager.h"
 #include "webrtc/base/gunit.h"
+#include "webrtc/media/base/fakemediaengine.h"
+#include "webrtc/media/base/fakevideocapturer.h"
+#include "webrtc/media/base/fakevideorenderer.h"
+#include "webrtc/media/webrtc/webrtcvideoframe.h"
 
 using webrtc::FakeConstraints;
 using webrtc::VideoSource;

talk/app/webrtc/videosourceinterface.h

@@ -29,8 +29,8 @@
 #define TALK_APP_WEBRTC_VIDEOSOURCEINTERFACE_H_
 
 #include "talk/app/webrtc/mediastreaminterface.h"
-#include "talk/media/base/mediachannel.h"
+#include "webrtc/media/base/mediachannel.h"
-#include "talk/media/base/videorenderer.h"
+#include "webrtc/media/base/videorenderer.h"
 
 namespace webrtc {
 

talk/app/webrtc/videotrack_unittest.cc

@@ -31,11 +31,11 @@
 #include "talk/app/webrtc/test/fakevideotrackrenderer.h"
 #include "talk/app/webrtc/videosource.h"
 #include "talk/app/webrtc/videotrack.h"
-#include "talk/media/base/fakemediaengine.h"
-#include "talk/media/webrtc/webrtcvideoframe.h"
 #include "talk/session/media/channelmanager.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/media/base/fakemediaengine.h"
+#include "webrtc/media/webrtc/webrtcvideoframe.h"
 
 using webrtc::FakeVideoTrackRenderer;
 using webrtc::VideoSource;

talk/app/webrtc/videotrackrenderers.cc

@@ -26,7 +26,7 @@
  */
 
 #include "talk/app/webrtc/videotrackrenderers.h"
-#include "talk/media/webrtc/webrtcvideoframe.h"
+#include "webrtc/media/webrtc/webrtcvideoframe.h"
 
 namespace webrtc {
 

talk/app/webrtc/videotrackrenderers.h

@@ -31,9 +31,9 @@
 #include <set>
 
 #include "talk/app/webrtc/mediastreaminterface.h"
-#include "talk/media/base/videorenderer.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/media/base/videorenderer.h"
 
 namespace webrtc {
 

talk/app/webrtc/webrtcsdp.cc

@@ -27,29 +27,29 @@
 
 #include "talk/app/webrtc/webrtcsdp.h"
 
+#include <ctype.h>
 #include <limits.h>
 #include <stdio.h>
 #include <algorithm>
 #include <string>
 #include <vector>
-#include <ctype.h>
 
 #include "talk/app/webrtc/jsepicecandidate.h"
 #include "talk/app/webrtc/jsepsessiondescription.h"
-#include "talk/media/base/codec.h"
-#include "talk/media/base/constants.h"
-#include "talk/media/base/cryptoparams.h"
-#include "talk/media/base/rtputils.h"
-#include "talk/media/sctp/sctpdataengine.h"
-#include "webrtc/p2p/base/candidate.h"
-#include "webrtc/p2p/base/constants.h"
-#include "webrtc/p2p/base/port.h"
 #include "talk/session/media/mediasession.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/messagedigest.h"
 #include "webrtc/base/stringutils.h"
+#include "webrtc/media/base/codec.h"
+#include "webrtc/media/base/constants.h"
+#include "webrtc/media/base/cryptoparams.h"
+#include "webrtc/media/base/rtputils.h"
+#include "webrtc/media/sctp/sctpdataengine.h"
+#include "webrtc/p2p/base/candidate.h"
+#include "webrtc/p2p/base/constants.h"
+#include "webrtc/p2p/base/port.h"
 
 using cricket::AudioContentDescription;
 using cricket::Candidate;

talk/app/webrtc/webrtcsdp_unittest.cc

@@ -34,8 +34,6 @@
 #include "talk/app/webrtc/test/androidtestinitializer.h"
 #endif
 #include "talk/app/webrtc/webrtcsdp.h"
-#include "talk/media/base/constants.h"
-#include "webrtc/p2p/base/constants.h"
 #include "talk/session/media/mediasession.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
@@ -44,6 +42,8 @@
 #include "webrtc/base/sslfingerprint.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
+#include "webrtc/media/base/constants.h"
+#include "webrtc/p2p/base/constants.h"
 
 using cricket::AudioCodec;
 using cricket::AudioContentDescription;

talk/app/webrtc/webrtcsession.cc

@@ -40,8 +40,6 @@
 #include "talk/app/webrtc/peerconnectioninterface.h"
 #include "talk/app/webrtc/sctputils.h"
 #include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
-#include "talk/media/base/constants.h"
-#include "talk/media/base/videocapturer.h"
 #include "talk/session/media/channel.h"
 #include "talk/session/media/channelmanager.h"
 #include "talk/session/media/mediasession.h"
@@ -53,6 +51,8 @@
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/call.h"
+#include "webrtc/media/base/constants.h"
+#include "webrtc/media/base/videocapturer.h"
 #include "webrtc/p2p/base/portallocator.h"
 #include "webrtc/p2p/base/transportchannel.h"
 

talk/app/webrtc/webrtcsession.h

@@ -37,11 +37,11 @@
 #include "talk/app/webrtc/mediastreamprovider.h"
 #include "talk/app/webrtc/peerconnectioninterface.h"
 #include "talk/app/webrtc/statstypes.h"
-#include "talk/media/base/mediachannel.h"
 #include "talk/session/media/mediasession.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/thread.h"
+#include "webrtc/media/base/mediachannel.h"
 #include "webrtc/p2p/base/transportcontroller.h"
 
 namespace cricket {

talk/app/webrtc/webrtcsession_unittest.cc

@@ -42,15 +42,6 @@
 #include "talk/app/webrtc/videotrack.h"
 #include "talk/app/webrtc/webrtcsession.h"
 #include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
-#include "talk/media/base/fakemediaengine.h"
-#include "talk/media/base/fakevideorenderer.h"
-#include "talk/media/base/mediachannel.h"
-#include "talk/media/webrtc/fakewebrtccall.h"
-#include "webrtc/p2p/base/stunserver.h"
-#include "webrtc/p2p/base/teststunserver.h"
-#include "webrtc/p2p/base/testturnserver.h"
-#include "webrtc/p2p/base/transportchannel.h"
-#include "webrtc/p2p/client/basicportallocator.h"
 #include "talk/session/media/channelmanager.h"
 #include "talk/session/media/mediasession.h"
 #include "webrtc/base/fakenetwork.h"
@@ -65,6 +56,15 @@
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/virtualsocketserver.h"
+#include "webrtc/media/base/fakemediaengine.h"
+#include "webrtc/media/base/fakevideorenderer.h"
+#include "webrtc/media/base/mediachannel.h"
+#include "webrtc/media/webrtc/fakewebrtccall.h"
+#include "webrtc/p2p/base/stunserver.h"
+#include "webrtc/p2p/base/teststunserver.h"
+#include "webrtc/p2p/base/testturnserver.h"
+#include "webrtc/p2p/base/transportchannel.h"
+#include "webrtc/p2p/client/basicportallocator.h"
 
 #define MAYBE_SKIP_TEST(feature)                    \
   if (!(feature())) {                               \

talk/app/webrtc/webrtcsessiondescriptionfactory.h

@@ -31,9 +31,9 @@
 #include "talk/app/webrtc/dtlsidentitystore.h"
 #include "talk/app/webrtc/peerconnectioninterface.h"
 #include "talk/session/media/mediasession.h"
-#include "webrtc/p2p/base/transportdescriptionfactory.h"
 #include "webrtc/base/messagehandler.h"
 #include "webrtc/base/rtccertificate.h"
+#include "webrtc/p2p/base/transportdescriptionfactory.h"
 
 namespace cricket {
 class ChannelManager;

talk/build/common.gypi

@@ -30,7 +30,6 @@
 {
   'variables': {
     'webrtc_root%': '<(DEPTH)/webrtc',
-    'libjingle_tests_additional_deps%': [],
     # TODO(ronghuawu): For now, disable the Chrome plugins, which causes a
     # flood of chromium-style warnings.
     'clang_use_chrome_plugins%': 0,
@@ -43,8 +42,6 @@
     'build_expat%': 1,
     'build_json%': 1,
     'build_libsrtp%': 1,
-    'build_libyuv%': 1,
-    'build_usrsctp%': 1,
     # Make it possible to provide custom locations for some libraries.
     'libyuv_dir%': '<(DEPTH)/third_party/libyuv',
 
@@ -65,8 +62,6 @@
       # Feature selection
       'HAVE_SCTP',
       'HAVE_SRTP',
-      'HAVE_WEBRTC_VIDEO',
-      'HAVE_WEBRTC_VOICE',
     ],
     'conditions': [
       ['OS=="linux"', {

talk/libjingle.gyp

@@ -439,273 +439,12 @@
   ],
 
   'targets': [
-    {
-      'target_name': 'libjingle',
-      'type': 'none',
-      'dependencies': [
-        '<(webrtc_root)/base/base.gyp:rtc_base',
-      ],
-      'conditions': [
-        ['build_json==1', {
-          'dependencies': [
-            '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
-          ],
-          'export_dependent_settings': [
-            '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
-          ],
-        }],
-        ['build_expat==1', {
-          'dependencies': [
-            '<(DEPTH)/third_party/expat/expat.gyp:expat',
-          ],
-          'export_dependent_settings': [
-            '<(DEPTH)/third_party/expat/expat.gyp:expat',
-          ],
-        }],
-      ],
-    },  # target libjingle
-    {
-      'target_name': 'libjingle_media',
-      'type': 'static_library',
-      'dependencies': [
-        '<(webrtc_root)/common.gyp:webrtc_common',
-        '<(webrtc_root)/modules/modules.gyp:video_render_module',
-        '<(webrtc_root)/webrtc.gyp:webrtc',
-        '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
-        '<(webrtc_root)/sound/sound.gyp:rtc_sound',
-        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
-        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-        '<(webrtc_root)/libjingle/xmllite/xmllite.gyp:rtc_xmllite',
-        '<(webrtc_root)/libjingle/xmpp/xmpp.gyp:rtc_xmpp',
-        '<(webrtc_root)/p2p/p2p.gyp:rtc_p2p',
-        'libjingle',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '<(libyuv_dir)/include',
-        ],
-      },
-      'sources': [
-        'media/base/audioframe.h',
-        'media/base/audiorenderer.h',
-        'media/base/capturemanager.cc',
-        'media/base/capturemanager.h',
-        'media/base/capturerenderadapter.cc',
-        'media/base/capturerenderadapter.h',
-        'media/base/codec.cc',
-        'media/base/codec.h',
-        'media/base/constants.cc',
-        'media/base/constants.h',
-        'media/base/cpuid.cc',
-        'media/base/cpuid.h',
-        'media/base/cryptoparams.h',
-        'media/base/device.h',
-        'media/base/fakescreencapturerfactory.h',
-        'media/base/hybriddataengine.h',
-        'media/base/mediachannel.h',
-        'media/base/mediacommon.h',
-        'media/base/mediaengine.cc',
-        'media/base/mediaengine.h',
-        'media/base/rtpdataengine.cc',
-        'media/base/rtpdataengine.h',
-        'media/base/rtpdump.cc',
-        'media/base/rtpdump.h',
-        'media/base/rtputils.cc',
-        'media/base/rtputils.h',
-        'media/base/screencastid.h',
-        'media/base/streamparams.cc',
-        'media/base/streamparams.h',
-        'media/base/turnutils.cc',
-        'media/base/turnutils.h',
-        'media/base/videoadapter.cc',
-        'media/base/videoadapter.h',
-        'media/base/videocapturer.cc',
-        'media/base/videocapturer.h',
-        'media/base/videocapturerfactory.h',
-        'media/base/videocommon.cc',
-        'media/base/videocommon.h',
-        'media/base/videoframe.cc',
-        'media/base/videoframe.h',
-        'media/base/videoframefactory.cc',
-        'media/base/videoframefactory.h',
-        'media/base/videorenderer.h',
-        'media/base/yuvframegenerator.cc',
-        'media/base/yuvframegenerator.h',
-        'media/devices/deviceinfo.h',
-        'media/devices/devicemanager.cc',
-        'media/devices/devicemanager.h',
-        'media/devices/dummydevicemanager.h',
-        'media/devices/filevideocapturer.cc',
-        'media/devices/filevideocapturer.h',
-        'media/devices/videorendererfactory.h',
-        'media/devices/yuvframescapturer.cc',
-        'media/devices/yuvframescapturer.h',
-        'media/sctp/sctpdataengine.cc',
-        'media/sctp/sctpdataengine.h',
-        'media/webrtc/nullwebrtcvideoengine.h',
-        'media/webrtc/simulcast.cc',
-        'media/webrtc/simulcast.h',
-        'media/webrtc/webrtccommon.h',
-        'media/webrtc/webrtcmediaengine.cc',
-        'media/webrtc/webrtcmediaengine.h',
-        'media/webrtc/webrtcmediaengine.cc',
-        'media/webrtc/webrtcvideocapturer.cc',
-        'media/webrtc/webrtcvideocapturer.h',
-        'media/webrtc/webrtcvideocapturerfactory.h',
-        'media/webrtc/webrtcvideocapturerfactory.cc',
-        'media/webrtc/webrtcvideodecoderfactory.h',
-        'media/webrtc/webrtcvideoencoderfactory.h',
-        'media/webrtc/webrtcvideoengine2.cc',
-        'media/webrtc/webrtcvideoengine2.h',
-        'media/webrtc/webrtcvideoframe.cc',
-        'media/webrtc/webrtcvideoframe.h',
-        'media/webrtc/webrtcvideoframefactory.cc',
-        'media/webrtc/webrtcvideoframefactory.h',
-        'media/webrtc/webrtcvoe.h',
-        'media/webrtc/webrtcvoiceengine.cc',
-        'media/webrtc/webrtcvoiceengine.h',
-      ],
-      'conditions': [
-        ['build_libyuv==1', {
-          'dependencies': ['<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',],
-        }],
-        ['build_usrsctp==1', {
-          'include_dirs': [
-            # TODO(jiayl): move this into the direct_dependent_settings of
-            # usrsctp.gyp.
-            '<(DEPTH)/third_party/usrsctp/usrsctplib',
-          ],
-          'dependencies': [
-            '<(DEPTH)/third_party/usrsctp/usrsctp.gyp:usrsctplib',
-          ],
-        }],
-        ['build_with_chromium==1', {
-          'dependencies': [
-            '<(webrtc_root)/modules/modules.gyp:video_capture',
-            '<(webrtc_root)/modules/modules.gyp:video_render',
-          ],
-        }, {
-          'dependencies': [
-            '<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
-            '<(webrtc_root)/modules/modules.gyp:video_render_module_internal_impl',
-          ],
-        }],
-        ['OS=="linux"', {
-          'sources': [
-            'media/devices/libudevsymboltable.cc',
-            'media/devices/libudevsymboltable.h',
-            'media/devices/linuxdeviceinfo.cc',
-            'media/devices/linuxdevicemanager.cc',
-            'media/devices/linuxdevicemanager.h',
-            'media/devices/v4llookup.cc',
-            'media/devices/v4llookup.h',
-          ],
-          'conditions': [
-            ['use_gtk==1', {
-              'sources': [
-                'media/devices/gtkvideorenderer.cc',
-                'media/devices/gtkvideorenderer.h',
-              ],
-              'cflags': [
-                '<!@(pkg-config --cflags gobject-2.0 gthread-2.0 gtk+-2.0)',
-              ],
-            }],
-          ],
-          'include_dirs': [
-            'third_party/libudev'
-          ],
-          'libraries': [
-            '-lrt',
-          ],
-        }],
-        ['OS=="win"', {
-          'sources': [
-            'media/devices/gdivideorenderer.cc',
-            'media/devices/gdivideorenderer.h',
-            'media/devices/win32deviceinfo.cc',
-            'media/devices/win32devicemanager.cc',
-            'media/devices/win32devicemanager.h',
-          ],
-          'msvs_settings': {
-            'VCLibrarianTool': {
-              'AdditionalDependencies': [
-                'd3d9.lib',
-                'gdi32.lib',
-                'strmiids.lib',
-                'winmm.lib',
-              ],
-            },
-          },
-        }],
-        ['OS=="mac"', {
-          'sources': [
-            'media/devices/macdeviceinfo.cc',
-            'media/devices/macdevicemanager.cc',
-            'media/devices/macdevicemanager.h',
-            'media/devices/macdevicemanagermm.mm',
-          ],
-          'conditions': [
-            ['target_arch=="ia32"', {
-              'sources': [
-                'media/devices/carbonvideorenderer.cc',
-                'media/devices/carbonvideorenderer.h',
-              ],
-              'link_settings': {
-                'xcode_settings': {
-                  'OTHER_LDFLAGS': [
-                    '-framework Carbon',
-                  ],
-                },
-              },
-            }],
-          ],
-          'xcode_settings': {
-            'WARNING_CFLAGS': [
-              # TODO(ronghuawu): Update macdevicemanager.cc to stop using
-              # deprecated functions and remove this flag.
-              '-Wno-deprecated-declarations',
-            ],
-            # Disable partial availability warning to prevent errors
-            # in macdevicemanagermm.mm using AVFoundation.
-            # https://code.google.com/p/webrtc/issues/detail?id=4695
-            'WARNING_CFLAGS!': ['-Wpartial-availability'],
-          },
-          'link_settings': {
-            'xcode_settings': {
-              'OTHER_LDFLAGS': [
-                '-weak_framework AVFoundation',
-                '-framework Cocoa',
-                '-framework CoreAudio',
-                '-framework CoreVideo',
-                '-framework OpenGL',
-                '-framework QTKit',
-              ],
-            },
-          },
-        }],
-        ['OS=="ios"', {
-          'sources': [
-            'media/devices/mobiledevicemanager.cc',
-          ],
-          'include_dirs': [
-            # TODO(sjlee) Remove when vp8 is building for iOS.  vp8 pulls in
-            # libjpeg which pulls in libyuv which currently disabled.
-            '../third_party/libyuv/include',
-          ],
-        }],
-        ['OS=="android"', {
-          'sources': [
-            'media/devices/mobiledevicemanager.cc',
-          ],
-        }],
-      ],
-    },  # target libjingle_media
     {
       'target_name': 'libjingle_p2p',
       'type': 'static_library',
       'dependencies': [
-        'libjingle',
+        '<(webrtc_root)/base/base.gyp:rtc_base',
-        'libjingle_media',
+        '<(webrtc_root)/media/media.gyp:rtc_media',
       ],
       'conditions': [
         ['build_libsrtp==1', {
@@ -755,8 +494,8 @@
       'target_name': 'libjingle_peerconnection',
       'type': 'static_library',
       'dependencies': [
-        'libjingle',
+        '<(webrtc_root)/base/base.gyp:rtc_base',
-        'libjingle_media',
+        '<(webrtc_root)/media/media.gyp:rtc_media',
         'libjingle_p2p',
       ],
       'sources': [

talk/libjingle_media_unittest.isolate

@@ -1,47 +0,0 @@
-#
-# libjingle
-# Copyright 2013 Google Inc.
-#
-# Redistribution and use in source and binary forms, with or without
-# modification, are permitted provided that the following conditions are met:
-#
-#  1. Redistributions of source code must retain the above copyright notice,
-#     this list of conditions and the following disclaimer.
-#  2. Redistributions in binary form must reproduce the above copyright notice,
-#     this list of conditions and the following disclaimer in the documentation
-#     and/or other materials provided with the distribution.
-#  3. The name of the author may not be used to endorse or promote products
-#     derived from this software without specific prior written permission.
-#
-# THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
-# WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
-# MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
-# EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-# SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
-# OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
-# WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
-# OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
-# ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-
-{
-  'conditions': [
-    ['OS=="linux" or OS=="mac" or OS=="win"', {
-      'variables': {
-        'command': [
-          '<(PRODUCT_DIR)/libjingle_media_unittest<(EXECUTABLE_SUFFIX)',
-        ],
-        'files': [
-          '<(PRODUCT_DIR)/libjingle_media_unittest<(EXECUTABLE_SUFFIX)',
-          'media/testdata/captured-320x240-2s-48.frames',
-          'media/testdata/faces.1280x720_P420.yuv',
-          'media/testdata/faces_I420.jpg',
-          'media/testdata/faces_I422.jpg',
-          'media/testdata/faces_I444.jpg',
-          'media/testdata/faces_I411.jpg',
-          'media/testdata/faces_I400.jpg',
-        ],
-      },
-    }],
-  ],
-}

talk/libjingle_tests.gyp

@@ -27,136 +27,14 @@
 {
   'includes': ['build/common.gypi'],
   'targets': [
-    {
-      'target_name': 'libjingle_unittest_main',
-      'type': 'static_library',
-      'dependencies': [
-        '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
-        '<@(libjingle_tests_additional_deps)',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '<(libyuv_dir)/include',
-          '<(DEPTH)/testing/gtest/include',
-          '<(DEPTH)/testing/gtest',
-        ],
-      },
-      'conditions': [
-        ['build_libyuv==1', {
-          'dependencies': ['<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',],
-        }],
-      ],
-      'include_dirs': [
-         '<(DEPTH)/testing/gtest/include',
-         '<(DEPTH)/testing/gtest',
-       ],
-      'sources': [
-        'media/base/fakecapturemanager.h',
-        'media/base/fakemediaengine.h',
-        'media/base/fakenetworkinterface.h',
-        'media/base/fakertp.h',
-        'media/base/fakevideocapturer.h',
-        'media/base/fakevideorenderer.h',
-        'media/base/testutils.cc',
-        'media/base/testutils.h',
-        'media/devices/fakedevicemanager.h',
-        'media/webrtc/fakewebrtccall.cc',
-        'media/webrtc/fakewebrtccall.h',
-        'media/webrtc/fakewebrtccommon.h',
-        'media/webrtc/fakewebrtcdeviceinfo.h',
-        'media/webrtc/fakewebrtcvcmfactory.h',
-        'media/webrtc/fakewebrtcvideocapturemodule.h',
-        'media/webrtc/fakewebrtcvideoengine.h',
-        'media/webrtc/fakewebrtcvoiceengine.h',
-      ],
-    },  # target libjingle_unittest_main
-    {
-      'target_name': 'libjingle_media_unittest',
-      'type': 'executable',
-      'dependencies': [
-        '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
-        'libjingle.gyp:libjingle_media',
-        'libjingle_unittest_main',
-      ],
-      'sources': [
-        'media/base/capturemanager_unittest.cc',
-        'media/base/codec_unittest.cc',
-        'media/base/rtpdataengine_unittest.cc',
-        'media/base/rtpdump_unittest.cc',
-        'media/base/rtputils_unittest.cc',
-        'media/base/streamparams_unittest.cc',
-        'media/base/testutils.cc',
-        'media/base/testutils.h',
-        'media/base/turnutils_unittest.cc',
-        'media/base/videoadapter_unittest.cc',
-        'media/base/videocapturer_unittest.cc',
-        'media/base/videocommon_unittest.cc',
-        'media/base/videoengine_unittest.h',
-        'media/base/videoframe_unittest.h',
-        'media/devices/dummydevicemanager_unittest.cc',
-        'media/devices/filevideocapturer_unittest.cc',
-        'media/sctp/sctpdataengine_unittest.cc',
-        'media/webrtc/nullwebrtcvideoengine_unittest.cc',
-        'media/webrtc/simulcast_unittest.cc',
-        'media/webrtc/webrtcmediaengine_unittest.cc',
-        'media/webrtc/webrtcvideocapturer_unittest.cc',
-        'media/webrtc/webrtcvideoframe_unittest.cc',
-        'media/webrtc/webrtcvideoframefactory_unittest.cc',
-        # Disabled because some tests fail.
-        # TODO(ronghuawu): Reenable these tests.
-        # 'media/devices/devicemanager_unittest.cc',
-        'media/webrtc/webrtcvideoengine2_unittest.cc',
-        'media/webrtc/webrtcvoiceengine_unittest.cc',
-      ],
-      'conditions': [
-        ['OS=="win"', {
-          'conditions': [
-            ['use_openssl==0', {
-              'dependencies': [
-                '<(DEPTH)/net/third_party/nss/ssl.gyp:libssl',
-                '<(DEPTH)/third_party/nss/nss.gyp:nspr',
-                '<(DEPTH)/third_party/nss/nss.gyp:nss',
-              ],
-            }],
-          ],
-          'msvs_settings': {
-            'VCLinkerTool': {
-              'AdditionalDependencies': [
-                # TODO(ronghuawu): Since we've included strmiids in
-                # libjingle_media target, we shouldn't need this here.
-                # Find out why it doesn't work without this.
-                'strmiids.lib',
-              ],
-            },
-          },
-        }],
-        ['OS=="win" and clang==1', {
-          'msvs_settings': {
-            'VCCLCompilerTool': {
-              'AdditionalOptions': [
-                # Disable warnings failing when compiling with Clang on Windows.
-                # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
-                '-Wno-unused-function',
-              ],
-            },
-          },
-        },],
-        ['OS=="ios"', {
-          'sources!': [
-            'media/sctp/sctpdataengine_unittest.cc',
-          ],
-        }],
-      ],
-    },  # target libjingle_media_unittest
     {
       'target_name': 'libjingle_p2p_unittest',
       'type': 'executable',
       'dependencies': [
         '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
-        'libjingle.gyp:libjingle',
+        '<(webrtc_root)/webrtc.gyp:rtc_unittest_main',
         'libjingle.gyp:libjingle_peerconnection',
         'libjingle.gyp:libjingle_p2p',
-        'libjingle_unittest_main',
       ],
       'include_dirs': [
         '<(DEPTH)/third_party/libsrtp/srtp',
@@ -194,10 +72,9 @@
         '<(DEPTH)/testing/gmock.gyp:gmock',
         '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
         '<(webrtc_root)/common.gyp:webrtc_common',
-        'libjingle.gyp:libjingle',
+        '<(webrtc_root)/webrtc.gyp:rtc_unittest_main',
         'libjingle.gyp:libjingle_p2p',
         'libjingle.gyp:libjingle_peerconnection',
-        'libjingle_unittest_main',
       ],
       'direct_dependent_settings': {
         'include_dirs': [
@@ -426,19 +303,6 @@
     }],
     ['test_isolation_mode != "noop"', {
       'targets': [
-        {
-          'target_name': 'libjingle_media_unittest_run',
-          'type': 'none',
-          'dependencies': [
-            'libjingle_media_unittest',
-          ],
-          'includes': [
-            'build/isolate.gypi',
-          ],
-          'sources': [
-            'libjingle_media_unittest.isolate',
-          ],
-        },
         {
           'target_name': 'libjingle_p2p_unittest_run',
           'type': 'none',

talk/media/OWNERS

@@ -1,4 +0,0 @@
-mflodman@webrtc.org
-pbos@webrtc.org
-pthatcher@webrtc.org
-solenberg@webrtc.org

talk/session/media/audiomonitor.h

@@ -29,9 +29,9 @@
 #define TALK_SESSION_MEDIA_AUDIOMONITOR_H_
 
 #include <vector>
-#include "webrtc/p2p/base/port.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/thread.h"
+#include "webrtc/p2p/base/port.h"
 
 namespace cricket {
 

talk/session/media/bundlefilter.cc

@@ -27,8 +27,8 @@
 
 #include "talk/session/media/bundlefilter.h"
 
-#include "talk/media/base/rtputils.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/media/base/rtputils.h"
 
 namespace cricket {
 

talk/session/media/bundlefilter.h

@@ -33,8 +33,8 @@
 #include <set>
 #include <vector>
 
-#include "talk/media/base/streamparams.h"
 #include "webrtc/base/basictypes.h"
+#include "webrtc/media/base/streamparams.h"
 
 namespace cricket {
 

talk/session/media/channel.cc

@@ -29,8 +29,6 @@
 
 #include "talk/session/media/channel.h"
 
-#include "talk/media/base/constants.h"
-#include "talk/media/base/rtputils.h"
 #include "talk/session/media/channelmanager.h"
 #include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/bind.h"
@@ -40,6 +38,8 @@
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
+#include "webrtc/media/base/constants.h"
+#include "webrtc/media/base/rtputils.h"
 #include "webrtc/p2p/base/transportchannel.h"
 
 namespace cricket {

talk/session/media/channel.h

@@ -28,16 +28,12 @@
 #ifndef TALK_SESSION_MEDIA_CHANNEL_H_
 #define TALK_SESSION_MEDIA_CHANNEL_H_
 
-#include <string>
-#include <vector>
 #include <map>
 #include <set>
+#include <string>
 #include <utility>
+#include <vector>
 
-#include "talk/media/base/mediachannel.h"
-#include "talk/media/base/mediaengine.h"
-#include "talk/media/base/streamparams.h"
-#include "talk/media/base/videocapturer.h"
 #include "talk/session/media/audiomonitor.h"
 #include "talk/session/media/bundlefilter.h"
 #include "talk/session/media/mediamonitor.h"
@@ -50,6 +46,10 @@
 #include "webrtc/base/network.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/window.h"
+#include "webrtc/media/base/mediachannel.h"
+#include "webrtc/media/base/mediaengine.h"
+#include "webrtc/media/base/streamparams.h"
+#include "webrtc/media/base/videocapturer.h"
 #include "webrtc/media/base/videosinkinterface.h"
 #include "webrtc/p2p/base/transportcontroller.h"
 #include "webrtc/p2p/client/socketmonitor.h"

talk/session/media/channel_unittest.cc

@@ -25,14 +25,6 @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#include "talk/media/base/fakemediaengine.h"
-#include "talk/media/base/fakertp.h"
-#include "talk/media/base/fakescreencapturerfactory.h"
-#include "talk/media/base/fakevideocapturer.h"
-#include "talk/media/base/mediachannel.h"
-#include "talk/media/base/rtpdump.h"
-#include "talk/media/base/screencastid.h"
-#include "talk/media/base/testutils.h"
 #include "talk/session/media/channel.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/fileutils.h"
@@ -44,6 +36,14 @@
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/window.h"
+#include "webrtc/media/base/fakemediaengine.h"
+#include "webrtc/media/base/fakertp.h"
+#include "webrtc/media/base/fakescreencapturerfactory.h"
+#include "webrtc/media/base/fakevideocapturer.h"
+#include "webrtc/media/base/mediachannel.h"
+#include "webrtc/media/base/rtpdump.h"
+#include "webrtc/media/base/screencastid.h"
+#include "webrtc/media/base/testutils.h"
 #include "webrtc/p2p/base/faketransportcontroller.h"
 
 #define MAYBE_SKIP_TEST(feature)                    \

talk/session/media/channelmanager.cc

@@ -34,13 +34,13 @@
 #include <algorithm>
 
 #include "talk/app/webrtc/mediacontroller.h"
-#include "talk/media/base/capturemanager.h"
+#include "webrtc/media/base/capturemanager.h"
-#include "talk/media/base/device.h"
+#include "webrtc/media/base/device.h"
-#include "talk/media/base/hybriddataengine.h"
+#include "webrtc/media/base/hybriddataengine.h"
-#include "talk/media/base/rtpdataengine.h"
+#include "webrtc/media/base/rtpdataengine.h"
-#include "talk/media/base/videocapturer.h"
+#include "webrtc/media/base/videocapturer.h"
 #ifdef HAVE_SCTP
-#include "talk/media/sctp/sctpdataengine.h"
+#include "webrtc/media/sctp/sctpdataengine.h"
 #endif
 #include "talk/session/media/srtpfilter.h"
 #include "webrtc/base/bind.h"

talk/session/media/channelmanager.h

@@ -31,13 +31,13 @@
 #include <string>
 #include <vector>
 
-#include "talk/media/base/capturemanager.h"
-#include "talk/media/base/mediaengine.h"
 #include "talk/session/media/voicechannel.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/sigslotrepeater.h"
 #include "webrtc/base/thread.h"
+#include "webrtc/media/base/capturemanager.h"
+#include "webrtc/media/base/mediaengine.h"
 
 namespace webrtc {
 class MediaControllerInterface;

talk/session/media/channelmanager_unittest.cc

@@ -26,15 +26,15 @@
  */
 
 #include "talk/app/webrtc/fakemediacontroller.h"
-#include "talk/media/base/fakecapturemanager.h"
-#include "talk/media/base/fakemediaengine.h"
-#include "talk/media/base/fakevideocapturer.h"
-#include "talk/media/base/testutils.h"
-#include "talk/media/webrtc/fakewebrtccall.h"
 #include "talk/session/media/channelmanager.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/thread.h"
+#include "webrtc/media/base/fakecapturemanager.h"
+#include "webrtc/media/base/fakemediaengine.h"
+#include "webrtc/media/base/fakevideocapturer.h"
+#include "webrtc/media/base/testutils.h"
+#include "webrtc/media/webrtc/fakewebrtccall.h"
 #include "webrtc/p2p/base/faketransportcontroller.h"
 
 namespace cricket {

talk/session/media/currentspeakermonitor.cc

@@ -27,9 +27,9 @@
 
 #include "talk/session/media/currentspeakermonitor.h"
 
-#include "talk/media/base/streamparams.h"
 #include "talk/session/media/audiomonitor.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/media/base/streamparams.h"
 
 namespace cricket {
 

talk/session/media/externalhmac.cc

@@ -33,11 +33,11 @@
 
 extern "C" {
 #ifdef SRTP_RELATIVE_PATH
-#include "srtp.h"  // NOLINT
 #include "crypto_kernel.h"  // NOLINT
+#include "srtp.h"  // NOLINT
 #else
-#include "third_party/libsrtp/srtp/include/srtp.h"
 #include "third_party/libsrtp/srtp/crypto/include/crypto_kernel.h"
+#include "third_party/libsrtp/srtp/include/srtp.h"
 #endif  // SRTP_RELATIVE_PATH
 }
 

talk/session/media/mediamonitor.h

@@ -30,11 +30,11 @@
 #ifndef TALK_SESSION_MEDIA_MEDIAMONITOR_H_
 #define TALK_SESSION_MEDIA_MEDIAMONITOR_H_
 
-#include "talk/media/base/mediachannel.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/thread_annotations.h"
+#include "webrtc/media/base/mediachannel.h"
 
 namespace cricket {
 

talk/session/media/mediasession.cc

@@ -32,18 +32,18 @@
 #include <set>
 #include <utility>
 
-#include "talk/media/base/constants.h"
-#include "talk/media/base/cryptoparams.h"
 #include "talk/session/media/channelmanager.h"
 #include "talk/session/media/srtpfilter.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/stringutils.h"
+#include "webrtc/media/base/constants.h"
+#include "webrtc/media/base/cryptoparams.h"
 #include "webrtc/p2p/base/constants.h"
 
 #ifdef HAVE_SCTP
-#include "talk/media/sctp/sctpdataengine.h"
+#include "webrtc/media/sctp/sctpdataengine.h"
 #else
 static const uint32_t kMaxSctpSid = 1023;
 #endif

talk/session/media/mediasession.h

@@ -34,16 +34,16 @@
 #include <string>
 #include <vector>
 
-#include "talk/media/base/codec.h"
+#include "webrtc/base/scoped_ptr.h"
-#include "talk/media/base/constants.h"
+#include "webrtc/media/base/codec.h"
-#include "talk/media/base/cryptoparams.h"
+#include "webrtc/media/base/constants.h"
-#include "talk/media/base/mediachannel.h"
+#include "webrtc/media/base/cryptoparams.h"
-#include "talk/media/base/mediaengine.h"  // For DataChannelType
+#include "webrtc/media/base/mediachannel.h"
-#include "talk/media/base/streamparams.h"
+#include "webrtc/media/base/mediaengine.h"  // For DataChannelType
+#include "webrtc/media/base/streamparams.h"
 #include "webrtc/p2p/base/sessiondescription.h"
 #include "webrtc/p2p/base/transport.h"
 #include "webrtc/p2p/base/transportdescriptionfactory.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace cricket {
 

talk/session/media/mediasession_unittest.cc

@@ -28,17 +28,17 @@
 #include <string>
 #include <vector>
 
-#include "talk/media/base/codec.h"
-#include "talk/media/base/testutils.h"
-#include "webrtc/p2p/base/constants.h"
-#include "webrtc/p2p/base/transportdescription.h"
-#include "webrtc/p2p/base/transportinfo.h"
 #include "talk/session/media/mediasession.h"
 #include "talk/session/media/srtpfilter.h"
 #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/transportdescription.h"
+#include "webrtc/p2p/base/transportinfo.h"
 
 #ifdef HAVE_SRTP
 #define ASSERT_CRYPTO(cd, s, cs) \

talk/session/media/planarfunctions_unittest.cc

@@ -32,11 +32,11 @@
 #include "libyuv/convert_from_argb.h"
 #include "libyuv/mjpeg_decoder.h"
 #include "libyuv/planar_functions.h"
-#include "talk/media/base/testutils.h"
-#include "talk/media/base/videocommon.h"
 #include "webrtc/base/flags.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/media/base/testutils.h"
+#include "webrtc/media/base/videocommon.h"
 
 // Undefine macros for the windows build.
 #undef max

talk/session/media/rtcpmuxfilter.h

@@ -28,8 +28,8 @@
 #ifndef TALK_SESSION_MEDIA_RTCPMUXFILTER_H_
 #define TALK_SESSION_MEDIA_RTCPMUXFILTER_H_
 
-#include "webrtc/p2p/base/sessiondescription.h"
 #include "webrtc/base/basictypes.h"
+#include "webrtc/p2p/base/sessiondescription.h"
 
 namespace cricket {