Add check_deps rules in DEPS files.
Add fine-grained check_deps rules for all of WebRTC.
This will help both maintaining sane dependencies and provides a way
to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
Example:
buildtools/checkdeps/graphdeps.py --root=. --format=png \
--out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
--excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
will produce a neat webrtc.png image showcasing the dependencies
(according to the DEPS file) for the bitrate_controller module.
Some dependencies are filtered out for readability.
BUG=webrtc:5623
TESTED=Passing runs using:
buildtools/checkdeps/checkdeps.py --root=. talk
buildtools/checkdeps/checkdeps.py --root=. webrtc
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1796413002 .
Cr-Commit-Position: refs/heads/master@{#12008}
diff --git a/DEPS b/DEPS
index 1ec0583..7f2225c 100644
--- a/DEPS
+++ b/DEPS
@@ -23,25 +23,6 @@
},
}
-# Define rules for which include paths are allowed in our source.
-include_rules = [
- # Base is only used to build Android APK tests and may not be referenced by
- # WebRTC production code.
- '-base',
- '-chromium',
- '+external/webrtc/webrtc', # Android platform build.
- '+gflags',
- '+libyuv',
- '+net',
- '+talk',
- '+testing',
- '+third_party',
- '+unicode',
- '+usrsctplib',
- '+webrtc',
- '+vpx',
-]
-
hooks = [
{
# Check for legacy named top-level dir (named 'trunk').
diff --git a/talk/app/webrtc/DEPS b/talk/app/webrtc/DEPS
new file mode 100644
index 0000000..69ecd02
--- /dev/null
+++ b/talk/app/webrtc/DEPS
@@ -0,0 +1,7 @@
+include_rules = [
+ "+talk/app/webrtc/objc",
+ "+webrtc/video_frame.h",
+ "+webrtc/api",
+ "+webrtc/base",
+ "+webrtc/media",
+]
diff --git a/webrtc/DEPS b/webrtc/DEPS
new file mode 100644
index 0000000..292c996
--- /dev/null
+++ b/webrtc/DEPS
@@ -0,0 +1,47 @@
+# Define rules for which include paths are allowed in our source.
+include_rules = [
+ # Base is only used to build Android APK tests and may not be referenced by
+ # WebRTC production code.
+ "-base",
+ "-chromium",
+ "+external/webrtc/webrtc", # Android platform build.
+ "+gflags",
+ "+libyuv",
+ "+testing",
+ "-webrtc", # Has to be disabled; otherwise all dirs below will be allowed.
+ # Individual headers that will be moved out of here, see webrtc:
+ "+webrtc/audio_receive_stream.h",
+ "+webrtc/audio_send_stream.h",
+ "+webrtc/audio_sink.h",
+ "+webrtc/audio_state.h",
+ "+webrtc/call.h",
+ "+webrtc/common.h",
+ "+webrtc/common_types.h",
+ "+webrtc/config.h",
+ "+webrtc/engine_configurations.h",
+ "+webrtc/frame_callback.h",
+ "+webrtc/stream.h",
+ "+webrtc/transport.h",
+ "+webrtc/typedefs.h",
+ "+webrtc/video_decoder.h",
+ "+webrtc/video_encoder.h",
+ "+webrtc/video_frame.h",
+ "+webrtc/video_receive_stream.h",
+ "+webrtc/video_renderer.h",
+ "+webrtc/video_send_stream.h",
+
+ "+webrtc/base",
+ "+webrtc/modules/include",
+ "+webrtc/test",
+ "+webrtc/tools",
+]
+
+# The below rules will be removed when webrtc: is fixed.
+specific_include_rules = {
+ "audio_send_stream\.h": [
+ "+webrtc/modules/audio_coding",
+ ],
+ "video_frame\.h": [
+ "+webrtc/common_video",
+ ],
+}
diff --git a/webrtc/api/DEPS b/webrtc/api/DEPS
new file mode 100644
index 0000000..956d4d9
--- /dev/null
+++ b/webrtc/api/DEPS
@@ -0,0 +1,23 @@
+include_rules = [
+ "+third_party/libyuv",
+ "+webrtc/base",
+ "+webrtc/common_video",
+ "+webrtc/media",
+ "+webrtc/p2p",
+ "+webrtc/pc",
+ "+webrtc/modules/audio_device",
+ "+webrtc/modules/rtp_rtcp",
+ "+webrtc/modules/video_coding",
+ "+webrtc/modules/video_render",
+ "+webrtc/system_wrappers",
+]
+
+specific_include_rules = {
+ "androidtestinitializer\.cc": [
+ "+base/android", # Allowed only for Android tests.
+ "+webrtc/voice_engine",
+ ],
+ "peerconnection_jni\.cc": [
+ "+webrtc/voice_engine",
+ ]
+}
diff --git a/webrtc/audio/DEPS b/webrtc/audio/DEPS
new file mode 100644
index 0000000..63711ab
--- /dev/null
+++ b/webrtc/audio/DEPS
@@ -0,0 +1,10 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/voice_engine",
+ "+webrtc/modules/bitrate_controller",
+ "+webrtc/modules/congestion_controller",
+ "+webrtc/modules/pacing",
+ "+webrtc/modules/remote_bitrate_estimator",
+ "+webrtc/modules/rtp_rtcp",
+ "+webrtc/system_wrappers",
+]
diff --git a/webrtc/base/DEPS b/webrtc/base/DEPS
new file mode 100644
index 0000000..add7f38
--- /dev/null
+++ b/webrtc/base/DEPS
@@ -0,0 +1,11 @@
+include_rules = [
+ "+json",
+ "+third_party/jsoncpp",
+ "+webrtc/system_wrappers",
+]
+
+specific_include_rules = {
+ "gunit_prod.h": [
+ "+gtest",
+ ],
+}
diff --git a/webrtc/call/DEPS b/webrtc/call/DEPS
new file mode 100644
index 0000000..0f90308
--- /dev/null
+++ b/webrtc/call/DEPS
@@ -0,0 +1,13 @@
+include_rules = [
+ "+webrtc/audio",
+ "+webrtc/base",
+ "+webrtc/modules/audio_coding",
+ "+webrtc/modules/bitrate_controller",
+ "+webrtc/modules/congestion_controller",
+ "+webrtc/modules/pacing",
+ "+webrtc/modules/rtp_rtcp",
+ "+webrtc/modules/utility",
+ "+webrtc/system_wrappers",
+ "+webrtc/voice_engine",
+ "+webrtc/video",
+]
diff --git a/webrtc/common_audio/DEPS b/webrtc/common_audio/DEPS
new file mode 100644
index 0000000..2d21857
--- /dev/null
+++ b/webrtc/common_audio/DEPS
@@ -0,0 +1,5 @@
+include_rules = [
+ "+third_party/openmax_dl",
+ "+webrtc/base",
+ "+webrtc/system_wrappers",
+]
diff --git a/webrtc/common_audio/real_fourier_openmax.cc b/webrtc/common_audio/real_fourier_openmax.cc
index bc3e734..e560b6c 100644
--- a/webrtc/common_audio/real_fourier_openmax.cc
+++ b/webrtc/common_audio/real_fourier_openmax.cc
@@ -12,7 +12,7 @@
#include <cstdlib>
-#include "dl/sp/api/omxSP.h"
+#include "third_party/openmax_dl/dl/sp/api/omxSP.h"
#include "webrtc/base/checks.h"
namespace webrtc {
diff --git a/webrtc/common_video/DEPS b/webrtc/common_video/DEPS
new file mode 100644
index 0000000..2805958
--- /dev/null
+++ b/webrtc/common_video/DEPS
@@ -0,0 +1,4 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/system_wrappers",
+]
diff --git a/webrtc/examples/DEPS b/webrtc/examples/DEPS
new file mode 100644
index 0000000..a562df9
--- /dev/null
+++ b/webrtc/examples/DEPS
@@ -0,0 +1,7 @@
+include_rules = [
+ "+webrtc/api",
+ "+webrtc/base",
+ "+webrtc/media",
+ "+webrtc/modules/audio_device",
+ "+webrtc/p2p",
+]
diff --git a/webrtc/libjingle/DEPS b/webrtc/libjingle/DEPS
new file mode 100644
index 0000000..7f49277
--- /dev/null
+++ b/webrtc/libjingle/DEPS
@@ -0,0 +1,5 @@
+include_rules = [
+ "+third_party/expat",
+ "+webrtc/base",
+ "+webrtc/p2p",
+]
diff --git a/webrtc/libjingle/xmpp/chatroommodule_unittest.cc b/webrtc/libjingle/xmpp/chatroommodule_unittest.cc
deleted file mode 100644
index 65d2827..0000000
--- a/webrtc/libjingle/xmpp/chatroommodule_unittest.cc
+++ /dev/null
@@ -1,280 +0,0 @@
-/*
- * 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 <iostream>
-#include <sstream>
-#include <string>
-#include "buzz/chatroommodule.h"
-#include "buzz/constants.h"
-#include "buzz/xmlelement.h"
-#include "buzz/xmppengine.h"
-#include "common/common.h"
-#include "engine/util_unittest.h"
-#include "test/unittest-inl.h"
-#include "test/unittest.h"
-
-#define TEST_OK(x) TEST_EQ((x),XMPP_RETURN_OK)
-#define TEST_BADARGUMENT(x) TEST_EQ((x),XMPP_RETURN_BADARGUMENT)
-
-namespace buzz {
-
-class MultiUserChatModuleTest;
-
-static void
-WriteEnteredStatus(std::ostream& os, XmppChatroomEnteredStatus status) {
- switch(status) {
- case XMPP_CHATROOM_ENTERED_SUCCESS:
- os<<"success";
- break;
- case XMPP_CHATROOM_ENTERED_FAILURE_NICKNAME_CONFLICT:
- os<<"failure(nickname conflict)";
- break;
- case XMPP_CHATROOM_ENTERED_FAILURE_PASSWORD_REQUIRED:
- os<<"failure(password required)";
- break;
- case XMPP_CHATROOM_ENTERED_FAILURE_PASSWORD_INCORRECT:
- os<<"failure(password incorrect)";
- break;
- case XMPP_CHATROOM_ENTERED_FAILURE_NOT_A_MEMBER:
- os<<"failure(not a member)";
- break;
- case XMPP_CHATROOM_ENTERED_FAILURE_MEMBER_BANNED:
- os<<"failure(member banned)";
- break;
- case XMPP_CHATROOM_ENTERED_FAILURE_MAX_USERS:
- os<<"failure(max users)";
- break;
- case XMPP_CHATROOM_ENTERED_FAILURE_ROOM_LOCKED:
- os<<"failure(room locked)";
- break;
- case XMPP_CHATROOM_ENTERED_FAILURE_UNSPECIFIED:
- os<<"failure(unspecified)";
- break;
- default:
- os<<"unknown";
- break;
- }
-}
-
-static void
-WriteExitedStatus(std::ostream& os, XmppChatroomExitedStatus status) {
- switch (status) {
- case XMPP_CHATROOM_EXITED_REQUESTED:
- os<<"requested";
- break;
- case XMPP_CHATROOM_EXITED_BANNED:
- os<<"banned";
- break;
- case XMPP_CHATROOM_EXITED_KICKED:
- os<<"kicked";
- break;
- case XMPP_CHATROOM_EXITED_NOT_A_MEMBER:
- os<<"not member";
- break;
- case XMPP_CHATROOM_EXITED_SYSTEM_SHUTDOWN:
- os<<"system shutdown";
- break;
- case XMPP_CHATROOM_EXITED_UNSPECIFIED:
- os<<"unspecified";
- break;
- default:
- os<<"unknown";
- break;
- }
-}
-
-//! This session handler saves all calls to a string. These are events and
-//! data delivered form the engine to application code.
-class XmppTestChatroomHandler : public XmppChatroomHandler {
-public:
- XmppTestChatroomHandler() {}
- virtual ~XmppTestChatroomHandler() {}
-
- void ChatroomEnteredStatus(XmppChatroomModule* room,
- XmppChatroomEnteredStatus status) {
- RTC_UNUSED(room);
- ss_ <<"[ChatroomEnteredStatus status: ";
- WriteEnteredStatus(ss_, status);
- ss_ <<"]";
- }
-
-
- void ChatroomExitedStatus(XmppChatroomModule* room,
- XmppChatroomExitedStatus status) {
- RTC_UNUSED(room);
- ss_ <<"[ChatroomExitedStatus status: ";
- WriteExitedStatus(ss_, status);
- ss_ <<"]";
- }
-
- void MemberEntered(XmppChatroomModule* room,
- const XmppChatroomMember* entered_member) {
- RTC_UNUSED(room);
- ss_ << "[MemberEntered " << entered_member->member_jid().Str() << "]";
- }
-
- void MemberExited(XmppChatroomModule* room,
- const XmppChatroomMember* exited_member) {
- RTC_UNUSED(room);
- ss_ << "[MemberExited " << exited_member->member_jid().Str() << "]";
- }
-
- void MemberChanged(XmppChatroomModule* room,
- const XmppChatroomMember* changed_member) {
- RTC_UNUSED(room);
- ss_ << "[MemberChanged " << changed_member->member_jid().Str() << "]";
- }
-
- virtual void MessageReceived(XmppChatroomModule* room, const XmlElement& message) {
- RTC_UNUSED2(room, message);
- }
-
-
- std::string Str() {
- return ss_.str();
- }
-
- std::string StrClear() {
- std::string result = ss_.str();
- ss_.str("");
- return result;
- }
-
-private:
- std::stringstream ss_;
-};
-
-//! This is the class that holds all of the unit test code for the
-//! roster module
-class XmppChatroomModuleTest : public UnitTest {
-public:
- XmppChatroomModuleTest() {}
-
- void TestEnterExitChatroom() {
- std::stringstream dump;
-
- // Configure the engine
- rtc::scoped_ptr<XmppEngine> engine(XmppEngine::Create());
- XmppTestHandler handler(engine.get());
-
- // Configure the module and handler
- rtc::scoped_ptr<XmppChatroomModule> chatroom(XmppChatroomModule::Create());
-
- // Configure the module handler
- chatroom->RegisterEngine(engine.get());
-
- // Set up callbacks
- engine->SetOutputHandler(&handler);
- engine->AddStanzaHandler(&handler);
- engine->SetSessionHandler(&handler);
-
- // Set up minimal login info
- engine->SetUser(Jid("david@my-server"));
- engine->SetPassword("david");
-
- // Do the whole login handshake
- RunLogin(this, engine.get(), &handler);
- TEST_EQ("", handler.OutputActivity());
-
- // Get the chatroom and set the handler
- XmppTestChatroomHandler chatroom_handler;
- chatroom->set_chatroom_handler(static_cast<XmppChatroomHandler*>(&chatroom_handler));
-
- // try to enter the chatroom
- TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_NOT_IN_ROOM);
- chatroom->set_nickname("thirdwitch");
- chatroom->set_chatroom_jid(Jid("darkcave@my-server"));
- chatroom->RequestEnterChatroom("", XMPP_CONNECTION_STATUS_UNKNOWN, "en");
- TEST_EQ(chatroom_handler.StrClear(), "");
- TEST_EQ(handler.OutputActivity(),
- "<presence to=\"darkcave@my-server/thirdwitch\">"
- "<muc:x xmlns:muc=\"http://jabber.org/protocol/muc\"/>"
- "</presence>");
- TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_REQUESTED_ENTER);
-
- // simulate the server and test the client
- std::string input;
- input = "<presence from=\"darkcave@my-server/firstwitch\" to=\"david@my-server\">"
- "<x xmlns=\"http://jabber.org/protocol/muc#user\">"
- "<item affiliation=\"owner\" role=\"participant\"/>"
- "</x>"
- "</presence>";
- TEST_OK(engine->HandleInput(input.c_str(), input.length()));
- TEST_EQ(chatroom_handler.StrClear(), "");
- TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_REQUESTED_ENTER);
-
- input = "<presence from=\"darkcave@my-server/secondwitch\" to=\"david@my-server\">"
- "<x xmlns=\"http://jabber.org/protocol/muc#user\">"
- "<item affiliation=\"member\" role=\"participant\"/>"
- "</x>"
- "</presence>";
- TEST_OK(engine->HandleInput(input.c_str(), input.length()));
- TEST_EQ(chatroom_handler.StrClear(), "");
- TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_REQUESTED_ENTER);
-
- input = "<presence from=\"darkcave@my-server/thirdwitch\" to=\"david@my-server\">"
- "<x xmlns=\"http://jabber.org/protocol/muc#user\">"
- "<item affiliation=\"member\" role=\"participant\"/>"
- "</x>"
- "</presence>";
- TEST_OK(engine->HandleInput(input.c_str(), input.length()));
- TEST_EQ(chatroom_handler.StrClear(),
- "[ChatroomEnteredStatus status: success]");
- TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_IN_ROOM);
-
- // simulate somebody else entering the room after we entered
- input = "<presence from=\"darkcave@my-server/fourthwitch\" to=\"david@my-server\">"
- "<x xmlns=\"http://jabber.org/protocol/muc#user\">"
- "<item affiliation=\"member\" role=\"participant\"/>"
- "</x>"
- "</presence>";
- TEST_OK(engine->HandleInput(input.c_str(), input.length()));
- TEST_EQ(chatroom_handler.StrClear(), "[MemberEntered darkcave@my-server/fourthwitch]");
- TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_IN_ROOM);
-
- // simulate somebody else leaving the room after we entered
- input = "<presence from=\"darkcave@my-server/secondwitch\" to=\"david@my-server\" type=\"unavailable\">"
- "<x xmlns=\"http://jabber.org/protocol/muc#user\">"
- "<item affiliation=\"member\" role=\"participant\"/>"
- "</x>"
- "</presence>";
- TEST_OK(engine->HandleInput(input.c_str(), input.length()));
- TEST_EQ(chatroom_handler.StrClear(), "[MemberExited darkcave@my-server/secondwitch]");
- TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_IN_ROOM);
-
- // try to leave the room
- chatroom->RequestExitChatroom();
- TEST_EQ(chatroom_handler.StrClear(), "");
- TEST_EQ(handler.OutputActivity(),
- "<presence to=\"darkcave@my-server/thirdwitch\" type=\"unavailable\"/>");
- TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_REQUESTED_EXIT);
-
- // simulate the server and test the client
- input = "<presence from=\"darkcave@my-server/thirdwitch\" to=\"david@my-server\" type=\"unavailable\">"
- "<x xmlns=\"http://jabber.org/protocol/muc#user\">"
- "<item affiliation=\"member\" role=\"participant\"/>"
- "</x>"
- "</presence>";
- TEST_OK(engine->HandleInput(input.c_str(), input.length()));
- TEST_EQ(chatroom_handler.StrClear(),
- "[ChatroomExitedStatus status: requested]");
- TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_NOT_IN_ROOM);
- }
-
-};
-
-// A global function that creates the test suite for this set of tests.
-TestBase* ChatroomModuleTest_Create() {
- TestSuite* suite = new TestSuite("ChatroomModuleTest");
- ADD_TEST(suite, XmppChatroomModuleTest, TestEnterExitChatroom);
- return suite;
-}
-
-}
diff --git a/webrtc/media/DEPS b/webrtc/media/DEPS
new file mode 100644
index 0000000..10f88d0
--- /dev/null
+++ b/webrtc/media/DEPS
@@ -0,0 +1,22 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/call",
+ "+webrtc/common_video",
+ "+webrtc/modules/audio_coding",
+ "+webrtc/modules/audio_device",
+ "+webrtc/modules/audio_processing",
+ "+webrtc/modules/video_capture",
+ "+webrtc/modules/video_coding",
+ "+webrtc/p2p",
+ "+webrtc/pc",
+ "+webrtc/sound",
+ "+webrtc/system_wrappers",
+ "+webrtc/voice_engine",
+ "+usrsctplib",
+]
+
+specific_include_rules = {
+ "win32devicemanager\.cc": [
+ "+third_party/logitech/files/logitechquickcam.h",
+ ],
+}
diff --git a/webrtc/modules/audio_coding/DEPS b/webrtc/modules/audio_coding/DEPS
new file mode 100644
index 0000000..31aa1c2
--- /dev/null
+++ b/webrtc/modules/audio_coding/DEPS
@@ -0,0 +1,7 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/call",
+ "+webrtc/common_audio",
+ "+webrtc/audio_coding/neteq/neteq_unittest.pb.h", # Different path.
+ "+webrtc/system_wrappers",
+]
diff --git a/webrtc/modules/audio_conference_mixer/DEPS b/webrtc/modules/audio_conference_mixer/DEPS
new file mode 100644
index 0000000..2805958
--- /dev/null
+++ b/webrtc/modules/audio_conference_mixer/DEPS
@@ -0,0 +1,4 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/system_wrappers",
+]
diff --git a/webrtc/modules/audio_device/DEPS b/webrtc/modules/audio_device/DEPS
new file mode 100644
index 0000000..2f4a597
--- /dev/null
+++ b/webrtc/modules/audio_device/DEPS
@@ -0,0 +1,11 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/common_audio",
+ "+webrtc/system_wrappers",
+]
+
+specific_include_rules = {
+ "ensure_initialized\.cc": [
+ "+base/android",
+ ],
+}
diff --git a/webrtc/modules/audio_processing/DEPS b/webrtc/modules/audio_processing/DEPS
new file mode 100644
index 0000000..e9ac967
--- /dev/null
+++ b/webrtc/modules/audio_processing/DEPS
@@ -0,0 +1,14 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/common_audio",
+ "+webrtc/system_wrappers",
+]
+
+specific_include_rules = {
+ ".*test\.cc": [
+ "+webrtc/tools",
+ # Android platform build has different paths.
+ "+gtest",
+ "+external/webrtc",
+ ],
+}
diff --git a/webrtc/modules/audio_processing/agc/agc_unittest.cc b/webrtc/modules/audio_processing/agc/agc_unittest.cc
index 25b99d8..8c6278f 100644
--- a/webrtc/modules/audio_processing/agc/agc_unittest.cc
+++ b/webrtc/modules/audio_processing/agc/agc_unittest.cc
@@ -10,8 +10,8 @@
#include "webrtc/modules/audio_processing/agc/agc.h"
-#include "gmock/gmock.h"
-#include "gtest/gtest.h"
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/test/testsupport/fileutils.h"
diff --git a/webrtc/modules/audio_processing/agc/mock_agc.h b/webrtc/modules/audio_processing/agc/mock_agc.h
index e362200..9e8f64e 100644
--- a/webrtc/modules/audio_processing/agc/mock_agc.h
+++ b/webrtc/modules/audio_processing/agc/mock_agc.h
@@ -13,7 +13,7 @@
#include "webrtc/modules/audio_processing/agc/agc.h"
-#include "gmock/gmock.h"
+#include "testing/gmock/include/gmock/gmock.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
diff --git a/webrtc/modules/bitrate_controller/DEPS b/webrtc/modules/bitrate_controller/DEPS
new file mode 100644
index 0000000..9a462b6
--- /dev/null
+++ b/webrtc/modules/bitrate_controller/DEPS
@@ -0,0 +1,5 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/call",
+ "+webrtc/system_wrappers",
+]
diff --git a/webrtc/modules/congestion_controller/DEPS b/webrtc/modules/congestion_controller/DEPS
new file mode 100644
index 0000000..d72e34d
--- /dev/null
+++ b/webrtc/modules/congestion_controller/DEPS
@@ -0,0 +1,5 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/system_wrappers",
+ "+webrtc/video",
+]
diff --git a/webrtc/modules/desktop_capture/DEPS b/webrtc/modules/desktop_capture/DEPS
new file mode 100644
index 0000000..2805958
--- /dev/null
+++ b/webrtc/modules/desktop_capture/DEPS
@@ -0,0 +1,4 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/system_wrappers",
+]
diff --git a/webrtc/modules/include/DEPS b/webrtc/modules/include/DEPS
new file mode 100644
index 0000000..aad6d8a
--- /dev/null
+++ b/webrtc/modules/include/DEPS
@@ -0,0 +1,4 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/common_video",
+]
diff --git a/webrtc/modules/media_file/DEPS b/webrtc/modules/media_file/DEPS
new file mode 100644
index 0000000..5c5452a
--- /dev/null
+++ b/webrtc/modules/media_file/DEPS
@@ -0,0 +1,5 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/common_audio",
+ "+webrtc/system_wrappers",
+]
diff --git a/webrtc/modules/pacing/DEPS b/webrtc/modules/pacing/DEPS
new file mode 100644
index 0000000..2805958
--- /dev/null
+++ b/webrtc/modules/pacing/DEPS
@@ -0,0 +1,4 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/system_wrappers",
+]
diff --git a/webrtc/modules/remote_bitrate_estimator/DEPS b/webrtc/modules/remote_bitrate_estimator/DEPS
new file mode 100644
index 0000000..9a863d9
--- /dev/null
+++ b/webrtc/modules/remote_bitrate_estimator/DEPS
@@ -0,0 +1,10 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/system_wrappers",
+]
+
+specific_include_rules = {
+ "nada\.h": [
+ "+webrtc/voice_engine",
+ ],
+}
diff --git a/webrtc/modules/rtp_rtcp/DEPS b/webrtc/modules/rtp_rtcp/DEPS
new file mode 100644
index 0000000..0720a15
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/DEPS
@@ -0,0 +1,6 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/call",
+ "+webrtc/common_video",
+ "+webrtc/system_wrappers",
+]
diff --git a/webrtc/modules/utility/DEPS b/webrtc/modules/utility/DEPS
new file mode 100644
index 0000000..1a2885b
--- /dev/null
+++ b/webrtc/modules/utility/DEPS
@@ -0,0 +1,6 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/common_audio",
+ "+webrtc/common_video",
+ "+webrtc/system_wrappers",
+]
diff --git a/webrtc/modules/video_capture/DEPS b/webrtc/modules/video_capture/DEPS
new file mode 100644
index 0000000..58ae9fe
--- /dev/null
+++ b/webrtc/modules/video_capture/DEPS
@@ -0,0 +1,5 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/common_video",
+ "+webrtc/system_wrappers",
+]
diff --git a/webrtc/modules/video_coding/DEPS b/webrtc/modules/video_coding/DEPS
new file mode 100644
index 0000000..512a0d8
--- /dev/null
+++ b/webrtc/modules/video_coding/DEPS
@@ -0,0 +1,9 @@
+include_rules = [
+ "+third_party/ffmpeg",
+ "+third_party/openh264",
+ "+vpx",
+ "+webrtc/base",
+ "+webrtc/common_video",
+ "+webrtc/system_wrappers",
+ "+webrtc/tools",
+]
diff --git a/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc b/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
index 3117e49..7be4eb1 100644
--- a/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
@@ -11,7 +11,7 @@
#include <memory>
#include <vector>
-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"
#include "vpx/vpx_encoder.h"
#include "vpx/vp8cx.h"
#include "webrtc/modules/video_coding/include/video_codec_interface.h"
diff --git a/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h b/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
index 9f0dc5c..2b2aa5d 100644
--- a/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
+++ b/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
@@ -15,6 +15,7 @@
#include <memory>
#include <vector>
+#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h"
@@ -22,8 +23,6 @@
#include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
#include "webrtc/video_frame.h"
-#include "gtest/gtest.h"
-
using ::testing::_;
using ::testing::AllOf;
using ::testing::Field;
diff --git a/webrtc/modules/video_processing/DEPS b/webrtc/modules/video_processing/DEPS
new file mode 100644
index 0000000..1a2885b
--- /dev/null
+++ b/webrtc/modules/video_processing/DEPS
@@ -0,0 +1,6 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/common_audio",
+ "+webrtc/common_video",
+ "+webrtc/system_wrappers",
+]
diff --git a/webrtc/modules/video_render/DEPS b/webrtc/modules/video_render/DEPS
new file mode 100644
index 0000000..58ae9fe
--- /dev/null
+++ b/webrtc/modules/video_render/DEPS
@@ -0,0 +1,5 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/common_video",
+ "+webrtc/system_wrappers",
+]
diff --git a/webrtc/p2p/DEPS b/webrtc/p2p/DEPS
new file mode 100644
index 0000000..161835f
--- /dev/null
+++ b/webrtc/p2p/DEPS
@@ -0,0 +1,5 @@
+include_rules = [
+ "+net",
+ "+webrtc/base",
+ "+webrtc/system_wrappers",
+]
diff --git a/webrtc/pc/DEPS b/webrtc/pc/DEPS
new file mode 100644
index 0000000..ca4f789
--- /dev/null
+++ b/webrtc/pc/DEPS
@@ -0,0 +1,13 @@
+include_rules = [
+ "+webrtc/api",
+ "+webrtc/base",
+ "+webrtc/media",
+ "+webrtc/p2p",
+ "+third_party/libsrtp"
+]
+
+specific_include_rules = {
+ "srtpfilter_unittest\.cc": [
+ "+crypto",
+ ],
+}
diff --git a/webrtc/sound/DEPS b/webrtc/sound/DEPS
new file mode 100644
index 0000000..7452a9f
--- /dev/null
+++ b/webrtc/sound/DEPS
@@ -0,0 +1,4 @@
+include_rules = [
+ "+webrtc/base",
+]
+
diff --git a/webrtc/system_wrappers/DEPS b/webrtc/system_wrappers/DEPS
new file mode 100644
index 0000000..7452a9f
--- /dev/null
+++ b/webrtc/system_wrappers/DEPS
@@ -0,0 +1,4 @@
+include_rules = [
+ "+webrtc/base",
+]
+
diff --git a/webrtc/test/DEPS b/webrtc/test/DEPS
new file mode 100644
index 0000000..27c0e74
--- /dev/null
+++ b/webrtc/test/DEPS
@@ -0,0 +1,13 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/call",
+ "+webrtc/common_video",
+ "+webrtc/modules/audio_coding",
+ "+webrtc/modules/audio_device",
+ "+webrtc/modules/media_file",
+ "+webrtc/modules/rtp_rtcp",
+ "+webrtc/modules/video_capture",
+ "+webrtc/modules/video_coding",
+ "+webrtc/system_wrappers",
+ "+webrtc/voice_engine",
+]
diff --git a/webrtc/tools/DEPS b/webrtc/tools/DEPS
new file mode 100644
index 0000000..73073f0
--- /dev/null
+++ b/webrtc/tools/DEPS
@@ -0,0 +1,8 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/common_video",
+ "+webrtc/modules/audio_processing",
+ "+webrtc/system_wrappers",
+ "+webrtc/voice_engine",
+]
+
diff --git a/webrtc/video/DEPS b/webrtc/video/DEPS
new file mode 100644
index 0000000..426f47c
--- /dev/null
+++ b/webrtc/video/DEPS
@@ -0,0 +1,17 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/call",
+ "+webrtc/common_video",
+ "+webrtc/modules/bitrate_controller",
+ "+webrtc/modules/congestion_controller",
+ "+webrtc/modules/pacing",
+ "+webrtc/modules/remote_bitrate_estimator",
+ "+webrtc/modules/rtp_rtcp",
+ "+webrtc/modules/utility",
+ "+webrtc/modules/video_coding",
+ "+webrtc/modules/video_capture",
+ "+webrtc/modules/video_processing",
+ "+webrtc/modules/video_render",
+ "+webrtc/system_wrappers",
+ "+webrtc/voice_engine",
+]
diff --git a/webrtc/voice_engine/DEPS b/webrtc/voice_engine/DEPS
new file mode 100644
index 0000000..224eeee
--- /dev/null
+++ b/webrtc/voice_engine/DEPS
@@ -0,0 +1,14 @@
+include_rules = [
+ "+webrtc/base",
+ "+webrtc/call",
+ "+webrtc/common_audio",
+ "+webrtc/modules/audio_coding",
+ "+webrtc/modules/audio_conference_mixer",
+ "+webrtc/modules/audio_device",
+ "+webrtc/modules/audio_processing",
+ "+webrtc/modules/media_file",
+ "+webrtc/modules/pacing",
+ "+webrtc/modules/rtp_rtcp",
+ "+webrtc/modules/utility",
+ "+webrtc/system_wrappers",
+]
