From 94a23f04afc715b42a1ac9d2abb8a2154af43f67 Mon Sep 17 00:00:00 2001 From: "kjellander@webrtc.org" Date: Thu, 17 Mar 2016 12:05:36 +0100 Subject: [PATCH] Reland "Add check_deps rules in DEPS files." Relanding https://codereview.webrtc.org/1796413002/ without the change to the openmax_dl include path (which broke downstream code). TBR=tommi@webrtc.org BUG=webrtc:5623 TESTED=Passing runs using: buildtools/checkdeps/checkdeps.py --root=. talk buildtools/checkdeps/checkdeps.py --root=. webrtc Review URL: https://codereview.webrtc.org/1804333002 . Cr-Commit-Position: refs/heads/master@{#12031} --- DEPS | 19 -- talk/app/webrtc/DEPS | 7 + webrtc/DEPS | 47 +++ webrtc/api/DEPS | 23 ++ webrtc/audio/DEPS | 10 + webrtc/base/DEPS | 11 + webrtc/call/DEPS | 13 + webrtc/common_audio/DEPS | 5 + webrtc/common_video/DEPS | 4 + webrtc/examples/DEPS | 8 + webrtc/libjingle/DEPS | 5 + .../libjingle/xmpp/chatroommodule_unittest.cc | 280 ------------------ webrtc/media/DEPS | 23 ++ webrtc/modules/audio_coding/DEPS | 7 + webrtc/modules/audio_conference_mixer/DEPS | 4 + webrtc/modules/audio_device/DEPS | 11 + webrtc/modules/audio_processing/DEPS | 14 + .../audio_processing/agc/agc_unittest.cc | 4 +- .../modules/audio_processing/agc/mock_agc.h | 2 +- webrtc/modules/bitrate_controller/DEPS | 5 + webrtc/modules/congestion_controller/DEPS | 5 + webrtc/modules/desktop_capture/DEPS | 4 + webrtc/modules/include/DEPS | 4 + webrtc/modules/media_file/DEPS | 5 + webrtc/modules/pacing/DEPS | 4 + webrtc/modules/remote_bitrate_estimator/DEPS | 10 + webrtc/modules/rtp_rtcp/DEPS | 6 + webrtc/modules/utility/DEPS | 6 + webrtc/modules/video_capture/DEPS | 5 + webrtc/modules/video_coding/DEPS | 9 + .../codecs/vp8/screenshare_layers_unittest.cc | 2 +- .../codecs/vp8/simulcast_unittest.h | 3 +- webrtc/modules/video_processing/DEPS | 6 + webrtc/modules/video_render/DEPS | 5 + webrtc/p2p/DEPS | 5 + webrtc/pc/DEPS | 13 + webrtc/sound/DEPS | 4 + webrtc/system_wrappers/DEPS | 4 + webrtc/test/DEPS | 13 + webrtc/tools/DEPS | 8 + webrtc/video/DEPS | 17 ++ webrtc/voice_engine/DEPS | 14 + 42 files changed, 349 insertions(+), 305 deletions(-) create mode 100644 talk/app/webrtc/DEPS create mode 100644 webrtc/DEPS create mode 100644 webrtc/api/DEPS create mode 100644 webrtc/audio/DEPS create mode 100644 webrtc/base/DEPS create mode 100644 webrtc/call/DEPS create mode 100644 webrtc/common_audio/DEPS create mode 100644 webrtc/common_video/DEPS create mode 100644 webrtc/examples/DEPS create mode 100644 webrtc/libjingle/DEPS delete mode 100644 webrtc/libjingle/xmpp/chatroommodule_unittest.cc create mode 100644 webrtc/media/DEPS create mode 100644 webrtc/modules/audio_coding/DEPS create mode 100644 webrtc/modules/audio_conference_mixer/DEPS create mode 100644 webrtc/modules/audio_device/DEPS create mode 100644 webrtc/modules/audio_processing/DEPS create mode 100644 webrtc/modules/bitrate_controller/DEPS create mode 100644 webrtc/modules/congestion_controller/DEPS create mode 100644 webrtc/modules/desktop_capture/DEPS create mode 100644 webrtc/modules/include/DEPS create mode 100644 webrtc/modules/media_file/DEPS create mode 100644 webrtc/modules/pacing/DEPS create mode 100644 webrtc/modules/remote_bitrate_estimator/DEPS create mode 100644 webrtc/modules/rtp_rtcp/DEPS create mode 100644 webrtc/modules/utility/DEPS create mode 100644 webrtc/modules/video_capture/DEPS create mode 100644 webrtc/modules/video_coding/DEPS create mode 100644 webrtc/modules/video_processing/DEPS create mode 100644 webrtc/modules/video_render/DEPS create mode 100644 webrtc/p2p/DEPS create mode 100644 webrtc/pc/DEPS create mode 100644 webrtc/sound/DEPS create mode 100644 webrtc/system_wrappers/DEPS create mode 100644 webrtc/test/DEPS create mode 100644 webrtc/tools/DEPS create mode 100644 webrtc/video/DEPS create mode 100644 webrtc/voice_engine/DEPS diff --git a/DEPS b/DEPS index 5aa8a4d305..ed33e9a317 100644 --- a/DEPS +++ b/DEPS @@ -23,25 +23,6 @@ deps_os = { }, } -# 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 0000000000..69ecd0279e --- /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 0000000000..292c99685b --- /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 0000000000..956d4d95c8 --- /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 0000000000..63711ab428 --- /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 0000000000..add7f38be5 --- /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 0000000000..0f9030853c --- /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 0000000000..7df03ea3fe --- /dev/null +++ b/webrtc/common_audio/DEPS @@ -0,0 +1,5 @@ +include_rules = [ + "+dl/sp/api", # For openmax_dl. + "+webrtc/base", + "+webrtc/system_wrappers", +] diff --git a/webrtc/common_video/DEPS b/webrtc/common_video/DEPS new file mode 100644 index 0000000000..2805958070 --- /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 0000000000..df9e0406b7 --- /dev/null +++ b/webrtc/examples/DEPS @@ -0,0 +1,8 @@ +include_rules = [ + "+webrtc/api", + "+webrtc/base", + "+webrtc/media", + "+webrtc/modules/audio_device", + "+webrtc/modules/video_capture", + "+webrtc/p2p", +] diff --git a/webrtc/libjingle/DEPS b/webrtc/libjingle/DEPS new file mode 100644 index 0000000000..7f492770e0 --- /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 65d28273cf..0000000000 --- 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 -#include -#include -#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 engine(XmppEngine::Create()); - XmppTestHandler handler(engine.get()); - - // Configure the module and handler - rtc::scoped_ptr 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(&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(), - "" - "" - ""); - TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_REQUESTED_ENTER); - - // simulate the server and test the client - std::string input; - input = "" - "" - "" - "" - ""; - TEST_OK(engine->HandleInput(input.c_str(), input.length())); - TEST_EQ(chatroom_handler.StrClear(), ""); - TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_REQUESTED_ENTER); - - input = "" - "" - "" - "" - ""; - TEST_OK(engine->HandleInput(input.c_str(), input.length())); - TEST_EQ(chatroom_handler.StrClear(), ""); - TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_REQUESTED_ENTER); - - input = "" - "" - "" - "" - ""; - 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 = "" - "" - "" - "" - ""; - 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 = "" - "" - "" - "" - ""; - 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(), - ""); - TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_REQUESTED_EXIT); - - // simulate the server and test the client - input = "" - "" - "" - "" - ""; - 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 0000000000..e44456317f --- /dev/null +++ b/webrtc/media/DEPS @@ -0,0 +1,23 @@ +include_rules = [ + "+webrtc/api", + "+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 0000000000..31aa1c25fb --- /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 0000000000..2805958070 --- /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 0000000000..2f4a597051 --- /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 0000000000..e9ac967c58 --- /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 25b99d8773..8c6278f40e 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 e362200d86..9e8f64e838 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 0000000000..9a462b6fc5 --- /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 0000000000..d72e34db6e --- /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 0000000000..2805958070 --- /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 0000000000..aad6d8a855 --- /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 0000000000..5c5452a0c0 --- /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 0000000000..2805958070 --- /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 0000000000..9a863d94a9 --- /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 0000000000..0720a15fec --- /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 0000000000..1a2885bbd3 --- /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 0000000000..58ae9fe714 --- /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 0000000000..512a0d8277 --- /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 3117e49788..7be4eb1bb2 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 #include -#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 9f0dc5ce93..2b2aa5de69 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 #include +#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 0000000000..1a2885bbd3 --- /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 0000000000..58ae9fe714 --- /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 0000000000..161835f343 --- /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 0000000000..ca4f789db9 --- /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 0000000000..7452a9fb16 --- /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 0000000000..7452a9fb16 --- /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 0000000000..27c0e744fd --- /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 0000000000..73073f02f4 --- /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 0000000000..426f47c423 --- /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 0000000000..224eeee676 --- /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", +]