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/talk/app/webrtc/webrtcsdp.cc b/talk/app/webrtc/webrtcsdp.cc
index f5fb7f7..16c0e69 100644
--- a/talk/app/webrtc/webrtcsdp.cc
+++ b/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;